#include <string>
#include <vector>
-#include "talk/base/basictypes.h"
-#include "talk/base/buffer.h"
-#include "talk/base/dscp.h"
-#include "talk/base/logging.h"
-#include "talk/base/sigslot.h"
-#include "talk/base/socket.h"
-#include "talk/base/window.h"
#include "talk/media/base/codec.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/dscp.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/socket.h"
+#include "webrtc/base/window.h"
// TODO(juberti): re-evaluate this include
#include "talk/session/media/audiomonitor.h"
-namespace talk_base {
+namespace rtc {
class Buffer;
class RateLimiter;
class Timing;
}
std::string ToString() const {
- return set_ ? talk_base::ToString(val_) : "";
+ return set_ ? rtc::ToString(val_) : "";
}
bool operator==(const Settable<T>& o) const {
video_start_bitrate.SetFrom(change.video_start_bitrate);
video_temporal_layer_screencast.SetFrom(
change.video_temporal_layer_screencast);
- video_temporal_layer_realtime.SetFrom(
- change.video_temporal_layer_realtime);
video_leaky_bucket.SetFrom(change.video_leaky_bucket);
video_highest_bitrate.SetFrom(change.video_highest_bitrate);
cpu_overuse_detection.SetFrom(change.cpu_overuse_detection);
system_high_adaptation_threshhold.SetFrom(
change.system_high_adaptation_threshhold);
buffered_mode_latency.SetFrom(change.buffered_mode_latency);
- lower_min_bitrate.SetFrom(change.lower_min_bitrate);
dscp.SetFrom(change.dscp);
suspend_below_min_bitrate.SetFrom(change.suspend_below_min_bitrate);
unsignalled_recv_stream_limit.SetFrom(change.unsignalled_recv_stream_limit);
use_simulcast_adapter.SetFrom(change.use_simulcast_adapter);
- skip_encoding_unused_streams.SetFrom(change.skip_encoding_unused_streams);
screencast_min_bitrate.SetFrom(change.screencast_min_bitrate);
use_improved_wifi_bandwidth_estimator.SetFrom(
change.use_improved_wifi_bandwidth_estimator);
video_high_bitrate == o.video_high_bitrate &&
video_start_bitrate == o.video_start_bitrate &&
video_temporal_layer_screencast == o.video_temporal_layer_screencast &&
- video_temporal_layer_realtime == o.video_temporal_layer_realtime &&
video_leaky_bucket == o.video_leaky_bucket &&
video_highest_bitrate == o.video_highest_bitrate &&
cpu_overuse_detection == o.cpu_overuse_detection &&
system_high_adaptation_threshhold ==
o.system_high_adaptation_threshhold &&
buffered_mode_latency == o.buffered_mode_latency &&
- lower_min_bitrate == o.lower_min_bitrate &&
dscp == o.dscp &&
suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit &&
use_simulcast_adapter == o.use_simulcast_adapter &&
- skip_encoding_unused_streams == o.skip_encoding_unused_streams &&
screencast_min_bitrate == o.screencast_min_bitrate &&
use_improved_wifi_bandwidth_estimator ==
o.use_improved_wifi_bandwidth_estimator &&
ost << ToStringIfSet("start bitrate", video_start_bitrate);
ost << ToStringIfSet("video temporal layer screencast",
video_temporal_layer_screencast);
- ost << ToStringIfSet("video temporal layer realtime",
- video_temporal_layer_realtime);
ost << ToStringIfSet("leaky bucket", video_leaky_bucket);
ost << ToStringIfSet("highest video bitrate", video_highest_bitrate);
ost << ToStringIfSet("cpu overuse detection", cpu_overuse_detection);
ost << ToStringIfSet("low", system_low_adaptation_threshhold);
ost << ToStringIfSet("high", system_high_adaptation_threshhold);
ost << ToStringIfSet("buffered mode latency", buffered_mode_latency);
- ost << ToStringIfSet("lower min bitrate", lower_min_bitrate);
ost << ToStringIfSet("dscp", dscp);
ost << ToStringIfSet("suspend below min bitrate",
suspend_below_min_bitrate);
ost << ToStringIfSet("num channels for early receive",
unsignalled_recv_stream_limit);
ost << ToStringIfSet("use simulcast adapter", use_simulcast_adapter);
- ost << ToStringIfSet("skip encoding unused streams",
- skip_encoding_unused_streams);
ost << ToStringIfSet("screencast min bitrate", screencast_min_bitrate);
ost << ToStringIfSet("improved wifi bwe",
use_improved_wifi_bandwidth_estimator);
Settable<int> video_start_bitrate;
// Experimental: Enable WebRTC layered screencast.
Settable<bool> video_temporal_layer_screencast;
- // Experimental: Enable WebRTC temporal layer strategy for realtime video.
- Settable<bool> video_temporal_layer_realtime;
// Enable WebRTC leaky bucket when sending media packets.
Settable<bool> video_leaky_bucket;
// Set highest bitrate mode for video.
SettablePercent system_high_adaptation_threshhold;
// Specify buffered mode latency in milliseconds.
Settable<int> buffered_mode_latency;
- // Make minimum configured send bitrate even lower than usual, at 30kbit.
- Settable<bool> lower_min_bitrate;
// Set DSCP value for packet sent from video channel.
Settable<bool> dscp;
// Enable WebRTC suspension of video. No video frames will be sent when the
Settable<int> unsignalled_recv_stream_limit;
// Enable use of simulcast adapter.
Settable<bool> use_simulcast_adapter;
- // Enables the encoder to skip encoding stream not actually sent due to too
- // low available bit rate.
- Settable<bool> skip_encoding_unused_streams;
// Force screencast to use a minimum bitrate
Settable<int> screencast_min_bitrate;
// Enable improved bandwidth estiamtor on wifi.
public:
enum SocketType { ST_RTP, ST_RTCP };
virtual bool SendPacket(
- talk_base::Buffer* packet,
- talk_base::DiffServCodePoint dscp = talk_base::DSCP_NO_CHANGE) = 0;
+ rtc::Buffer* packet,
+ rtc::DiffServCodePoint dscp = rtc::DSCP_NO_CHANGE) = 0;
virtual bool SendRtcp(
- talk_base::Buffer* packet,
- talk_base::DiffServCodePoint dscp = talk_base::DSCP_NO_CHANGE) = 0;
- virtual int SetOption(SocketType type, talk_base::Socket::Option opt,
+ rtc::Buffer* packet,
+ rtc::DiffServCodePoint dscp = rtc::DSCP_NO_CHANGE) = 0;
+ virtual int SetOption(SocketType type, rtc::Socket::Option opt,
int option) = 0;
virtual ~NetworkInterface() {}
};
// Sets the abstract interface class for sending RTP/RTCP data.
virtual void SetInterface(NetworkInterface *iface) {
- talk_base::CritScope cs(&network_interface_crit_);
+ rtc::CritScope cs(&network_interface_crit_);
network_interface_ = iface;
}
// Called when a RTP packet is received.
- virtual void OnPacketReceived(talk_base::Buffer* packet,
- const talk_base::PacketTime& packet_time) = 0;
+ virtual void OnPacketReceived(rtc::Buffer* packet,
+ const rtc::PacketTime& packet_time) = 0;
// Called when a RTCP packet is received.
- virtual void OnRtcpReceived(talk_base::Buffer* packet,
- const talk_base::PacketTime& packet_time) = 0;
+ virtual void OnRtcpReceived(rtc::Buffer* packet,
+ const rtc::PacketTime& packet_time) = 0;
// Called when the socket's ability to send has changed.
virtual void OnReadyToSend(bool ready) = 0;
// Creates a new outgoing media stream with SSRCs and CNAME as described
virtual bool SetMaxSendBandwidth(int bps) = 0;
// Base method to send packet using NetworkInterface.
- bool SendPacket(talk_base::Buffer* packet) {
+ bool SendPacket(rtc::Buffer* packet) {
return DoSendPacket(packet, false);
}
- bool SendRtcp(talk_base::Buffer* packet) {
+ bool SendRtcp(rtc::Buffer* packet) {
return DoSendPacket(packet, true);
}
int SetOption(NetworkInterface::SocketType type,
- talk_base::Socket::Option opt,
+ rtc::Socket::Option opt,
int option) {
- talk_base::CritScope cs(&network_interface_crit_);
+ rtc::CritScope cs(&network_interface_crit_);
if (!network_interface_)
return -1;
protected:
// This method sets DSCP |value| on both RTP and RTCP channels.
- int SetDscp(talk_base::DiffServCodePoint value) {
+ int SetDscp(rtc::DiffServCodePoint value) {
int ret;
ret = SetOption(NetworkInterface::ST_RTP,
- talk_base::Socket::OPT_DSCP,
+ rtc::Socket::OPT_DSCP,
value);
if (ret == 0) {
ret = SetOption(NetworkInterface::ST_RTCP,
- talk_base::Socket::OPT_DSCP,
+ rtc::Socket::OPT_DSCP,
value);
}
return ret;
}
private:
- bool DoSendPacket(talk_base::Buffer* packet, bool rtcp) {
- talk_base::CritScope cs(&network_interface_crit_);
+ bool DoSendPacket(rtc::Buffer* packet, bool rtcp) {
+ rtc::CritScope cs(&network_interface_crit_);
if (!network_interface_)
return false;
// |network_interface_| can be accessed from the worker_thread and
// from any MediaEngine threads. This critical section is to protect accessing
// of network_interface_ object.
- talk_base::CriticalSection network_interface_crit_;
+ rtc::CriticalSection network_interface_crit_;
NetworkInterface* network_interface_;
};
nominal_bitrate(0),
preferred_bitrate(0),
adapt_reason(0),
+ adapt_changes(0),
capture_jitter_ms(0),
avg_encode_ms(0),
encode_usage_percent(0),
int nominal_bitrate;
int preferred_bitrate;
int adapt_reason;
+ int adapt_changes;
int capture_jitter_ms;
int avg_encode_ms;
int encode_usage_percent;
virtual bool SendData(
const SendDataParams& params,
- const talk_base::Buffer& payload,
+ const rtc::Buffer& payload,
SendDataResult* result = NULL) = 0;
// Signals when data is received (params, data, len)
sigslot::signal3<const ReceiveDataParams&,
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