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* Update libjingle to 50389769.

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* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004

RISK=P1
TESTED=try bots
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org
2013-08-05 20:36:57 +00:00
parent f696f253b2
commit 9dba525627
54 changed files with 1797 additions and 518 deletions
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45
talk/media/webrtc/webrtcvoiceengine.h
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@ -275,7 +275,7 @@ template <class T, class E>
class WebRtcMediaChannel : public T, public webrtc::Transport {
public:
WebRtcMediaChannel(E *engine, int channel)
: engine_(engine), voe_channel_(channel), sequence_number_(-1) {}
: engine_(engine), voe_channel_(channel) {}
E *engine() { return engine_; }
int voe_channel() const { return voe_channel_; }
bool valid() const { return voe_channel_ != -1; }
@ -283,23 +283,10 @@ class WebRtcMediaChannel : public T, public webrtc::Transport {
protected:
// implements Transport interface
virtual int SendPacket(int channel, const void *data, int len) {
// We need to store the sequence number to be able to pick up
// the same sequence when the device is restarted.
// TODO(oja): Remove when WebRtc has fixed the problem.
int seq_num;
if (!GetRtpSeqNum(data, len, &seq_num)) {
return -1;
}
if (sequence_number() == -1) {
LOG(INFO) << "WebRtcVoiceMediaChannel sends first packet seqnum="
<< seq_num;
}
talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
if (!T::SendPacket(&packet)) {
return -1;
}
sequence_number_ = seq_num;
return len;
}
@ -308,14 +295,9 @@ class WebRtcMediaChannel : public T, public webrtc::Transport {
return T::SendRtcp(&packet) ? len : -1;
}
int sequence_number() const {
return sequence_number_;
}
private:
E *engine_;
int voe_channel_;
int sequence_number_;
};
// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
@ -393,16 +375,24 @@ class WebRtcVoiceMediaChannel
private:
struct WebRtcVoiceChannelInfo;
typedef std::map<uint32, WebRtcVoiceChannelInfo> ChannelMap;
void SetNack(uint32 ssrc, int channel, bool nack_enabled);
void SetNack(const ChannelMap& channels, bool nack_enabled);
bool SetSendCodec(const webrtc::CodecInst& send_codec);
bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
bool ChangePlayout(bool playout);
bool ChangeSend(SendFlags send);
bool ChangeSend(int channel, SendFlags send);
void ConfigureSendChannel(int channel);
bool DeleteChannel(int channel);
bool InConferenceMode() const {
return options_.conference_mode.GetWithDefaultIfUnset(false);
}
bool IsDefaultChannel(int channel_id) const {
return channel_id == voe_channel();
}
typedef std::map<uint32, WebRtcVoiceChannelInfo> ChannelMap;
talk_base::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
std::set<int> ringback_channels_; // channels playing ringback
std::vector<AudioCodec> recv_codecs_;
@ -415,17 +405,14 @@ class WebRtcVoiceMediaChannel
SendFlags desired_send_;
SendFlags send_;
// TODO(xians): Add support for multiple send channels.
uint32 send_ssrc_;
// Weak pointer to the renderer of the local audio track. It is owned by the
// track and will set to NULL when the track is going away or channel gets
// deleted. Used to notify the audio track that the media channel is added
// or removed.
AudioRenderer* local_renderer_;
// 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_;
uint32 default_receive_ssrc_;
// Note the default channel (voe_channel()) can reside in both
// receive_channels_ and send channel in non-conference mode and in that case
// it will only be there if a non-zero default_receive_ssrc_ is set.
// receive_channels_ and send_channels_ in non-conference mode and in that
// case it will only be there if a non-zero default_receive_ssrc_ is set.
ChannelMap receive_channels_; // for multiple sources
// receive_channels_ can be read from WebRtc callback thread. Access from
// the WebRtc thread must be synchronized with edits on the worker thread.