#include <string>
#include <vector>
-#include "talk/base/buffer.h"
-#include "talk/base/byteorder.h"
-#include "talk/base/logging.h"
-#include "talk/base/scoped_ptr.h"
-#include "talk/base/stream.h"
#include "talk/media/base/rtputils.h"
#include "talk/media/webrtc/webrtccommon.h"
#include "talk/media/webrtc/webrtcexport.h"
#include "talk/media/webrtc/webrtcvoe.h"
#include "talk/session/media/channel.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/byteorder.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/stream.h"
#include "webrtc/common.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#if !defined(LIBPEERCONNECTION_LIB) && \
!defined(LIBPEERCONNECTION_IMPLEMENTATION)
+// If you hit this, then you've tried to include this header from outside
+// the shared library. An instance of this class must only be created from
+// within the library that actually implements it. Otherwise use the
+// WebRtcMediaEngine to construct an instance.
#error "Bogus include."
#endif
+namespace webrtc {
+class VideoEngine;
+}
namespace cricket {
virtual int Rewind();
private:
- talk_base::MemoryStream mem_;
+ rtc::MemoryStream mem_;
bool loop_;
};
VoEWrapper* voe_wrapper_sc,
VoETraceWrapper* tracing);
~WebRtcVoiceEngine();
- bool Init(talk_base::Thread* worker_thread);
+ bool Init(rtc::Thread* worker_thread);
void Terminate();
int GetCapabilities();
webrtc::AudioDeviceModule* adm_sc);
// Starts AEC dump using existing file.
- bool StartAecDump(talk_base::PlatformFile file);
+ bool StartAecDump(rtc::PlatformFile file);
// Check whether the supplied trace should be ignored.
bool ShouldIgnoreTrace(const std::string& trace);
// allows us to selectively turn on and off different options easily
// at any time.
bool ApplyOptions(const AudioOptions& options);
- // Configure for using ACM2, if |enable| is true, otherwise configure for
- // ACM1.
- void EnableExperimentalAcm(bool enable);
virtual void Print(webrtc::TraceLevel level, const char* trace, int length);
virtual void CallbackOnError(int channel, int errCode);
// Given the device type, name, and id, find device id. Return true and
FrameSignal SignalRxMediaFrame;
FrameSignal SignalTxMediaFrame;
- static const int kDefaultLogSeverity = talk_base::LS_WARNING;
+ static const int kDefaultLogSeverity = rtc::LS_WARNING;
// The primary instance of WebRtc VoiceEngine.
- talk_base::scoped_ptr<VoEWrapper> voe_wrapper_;
+ rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
// A secondary instance, for playing out soundclips (on the 'ring' device).
- talk_base::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
+ rtc::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
bool voe_wrapper_sc_initialized_;
- talk_base::scoped_ptr<VoETraceWrapper> tracing_;
+ rtc::scoped_ptr<VoETraceWrapper> tracing_;
// The external audio device manager
webrtc::AudioDeviceModule* adm_;
webrtc::AudioDeviceModule* adm_sc_;
std::vector<AudioCodec> codecs_;
std::vector<RtpHeaderExtension> rtp_header_extensions_;
bool desired_local_monitor_enable_;
- talk_base::scoped_ptr<WebRtcMonitorStream> monitor_;
+ rtc::scoped_ptr<WebRtcMonitorStream> monitor_;
SoundclipList soundclips_;
ChannelList channels_;
// channels_ can be read from WebRtc callback thread. We need a lock on that
// callback as well as the RegisterChannel/UnregisterChannel.
- talk_base::CriticalSection channels_cs_;
+ rtc::CriticalSection channels_cs_;
webrtc::AgcConfig default_agc_config_;
webrtc::Config voe_config_;
- bool use_experimental_acm_;
bool initialized_;
// See SetOptions and SetOptionOverrides for a description of the
uint32 tx_processor_ssrc_;
uint32 rx_processor_ssrc_;
- talk_base::CriticalSection signal_media_critical_;
+ rtc::CriticalSection signal_media_critical_;
+
+ // Cache received experimental_aec and experimental_ns values, and apply them
+ // in case they are missing in the audio options. We need to do this because
+ // SetExtraOptions() will revert to defaults for options which are not
+ // provided.
+ Settable<bool> experimental_aec_;
+ Settable<bool> experimental_ns_;
};
// WebRtcMediaChannel is a class that implements the common WebRtc channel
protected:
// implements Transport interface
virtual int SendPacket(int channel, const void *data, int len) {
- talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
+ rtc::Buffer packet(data, len, kMaxRtpPacketLen);
if (!T::SendPacket(&packet)) {
return -1;
}
}
virtual int SendRTCPPacket(int channel, const void *data, int len) {
- talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
+ rtc::Buffer packet(data, len, kMaxRtpPacketLen);
return T::SendRtcp(&packet) ? len : -1;
}
virtual bool CanInsertDtmf();
virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags);
- virtual void OnPacketReceived(talk_base::Buffer* packet,
- const talk_base::PacketTime& packet_time);
- virtual void OnRtcpReceived(talk_base::Buffer* packet,
- const talk_base::PacketTime& packet_time);
+ virtual void OnPacketReceived(rtc::Buffer* packet,
+ const rtc::PacketTime& packet_time);
+ virtual void OnRtcpReceived(rtc::Buffer* packet,
+ const rtc::PacketTime& packet_time);
virtual void OnReadyToSend(bool ready) {}
virtual bool MuteStream(uint32 ssrc, bool on);
virtual bool SetStartSendBandwidth(int bps);
int GetReceiveChannelNum(uint32 ssrc);
int GetSendChannelNum(uint32 ssrc);
+ bool SetupSharedBandwidthEstimation(webrtc::VideoEngine* vie,
+ int vie_channel);
protected:
int GetLastEngineError() { return engine()->GetLastEngineError(); }
int GetOutputLevel(int channel);
bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
const RtpHeaderExtension* extension);
+ bool SetupSharedBweOnChannel(int voe_channel);
+
+ bool SetChannelRecvRtpHeaderExtensions(
+ int channel_id,
+ const std::vector<RtpHeaderExtension>& extensions);
+ bool SetChannelSendRtpHeaderExtensions(
+ int channel_id,
+ const std::vector<RtpHeaderExtension>& extensions);
- talk_base::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
+ rtc::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
std::set<int> ringback_channels_; // channels playing ringback
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
- talk_base::scoped_ptr<webrtc::CodecInst> send_codec_;
+ rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
bool send_bw_setting_;
int send_bw_bps_;
AudioOptions options_;
bool typing_noise_detected_;
SendFlags desired_send_;
SendFlags send_;
+ // shared_bwe_vie_ and shared_bwe_vie_channel_ together identifies a WebRTC
+ // VideoEngine channel that this voice channel should forward incoming packets
+ // to for Bandwidth Estimation purposes.
+ webrtc::VideoEngine* shared_bwe_vie_;
+ int shared_bwe_vie_channel_;
// send_channels_ contains the channels which are being used for sending.
// When the default channel (voe_channel) is used for sending, it is
// contained in send_channels_, otherwise not.
ChannelMap send_channels_;
+ std::vector<RtpHeaderExtension> send_extensions_;
uint32 default_receive_ssrc_;
// Note the default channel (voe_channel()) can reside in both
// receive_channels_ and send_channels_ in non-conference mode and in that
// the WebRtc thread must be synchronized with edits on the worker thread.
// Reads on the worker thread are ok.
//
+ std::vector<RtpHeaderExtension> receive_extensions_;
// Do not lock this on the VoE media processor thread; potential for deadlock
// exists.
- mutable talk_base::CriticalSection receive_channels_cs_;
+ mutable rtc::CriticalSection receive_channels_cs_;
};
} // namespace cricket
projects / platform / framework / web / crosswalk.git / blobdiff