Update libjingle to 57692857
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index 919248f..d7e7192 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -173,6 +173,7 @@
experimental_agc.SetFrom(change.experimental_agc);
experimental_aec.SetFrom(change.experimental_aec);
aec_dump.SetFrom(change.aec_dump);
+ experimental_acm.SetFrom(change.experimental_acm);
tx_agc_target_dbov.SetFrom(change.tx_agc_target_dbov);
tx_agc_digital_compression_gain.SetFrom(
change.tx_agc_digital_compression_gain);
@@ -200,6 +201,7 @@
experimental_aec == o.experimental_aec &&
adjust_agc_delta == o.adjust_agc_delta &&
aec_dump == o.aec_dump &&
+ experimental_acm == o.experimental_acm &&
tx_agc_target_dbov == o.tx_agc_target_dbov &&
tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
tx_agc_limiter == o.tx_agc_limiter &&
@@ -227,6 +229,7 @@
ost << ToStringIfSet("experimental_agc", experimental_agc);
ost << ToStringIfSet("experimental_aec", experimental_aec);
ost << ToStringIfSet("aec_dump", aec_dump);
+ ost << ToStringIfSet("experimental_acm", experimental_acm);
ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
ost << ToStringIfSet("tx_agc_digital_compression_gain",
tx_agc_digital_compression_gain);
@@ -263,6 +266,7 @@
Settable<bool> experimental_agc;
Settable<bool> experimental_aec;
Settable<bool> aec_dump;
+ Settable<bool> experimental_acm;
// Note that tx_agc_* only applies to non-experimental AGC.
Settable<uint16> tx_agc_target_dbov;
Settable<uint16> tx_agc_digital_compression_gain;
@@ -313,6 +317,7 @@
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);
}
bool operator==(const VideoOptions& o) const {
@@ -338,7 +343,8 @@
o.system_high_adaptation_threshhold &&
buffered_mode_latency == o.buffered_mode_latency &&
lower_min_bitrate == o.lower_min_bitrate &&
- dscp == o.dscp;
+ dscp == o.dscp &&
+ suspend_below_min_bitrate == o.suspend_below_min_bitrate;
}
std::string ToString() const {
@@ -367,6 +373,8 @@
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 << "}";
return ost.str();
}
@@ -415,6 +423,9 @@
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
+ // bitrate is below the configured minimum bitrate.
+ Settable<bool> suspend_below_min_bitrate;
};
// A class for playing out soundclips.
@@ -624,6 +635,35 @@
fraction_lost(0.0),
rtt_ms(0) {
}
+ void add_ssrc(const SsrcSenderInfo& stat) {
+ local_stats.push_back(stat);
+ }
+ // Temporary utility function for call sites that only provide SSRC.
+ // As more info is added into SsrcSenderInfo, this function should go away.
+ void add_ssrc(uint32 ssrc) {
+ SsrcSenderInfo stat;
+ stat.ssrc = ssrc;
+ add_ssrc(stat);
+ }
+ // Utility accessor for clients that are only interested in ssrc numbers.
+ std::vector<uint32> ssrcs() const {
+ std::vector<uint32> retval;
+ for (std::vector<SsrcSenderInfo>::const_iterator it = local_stats.begin();
+ it != local_stats.end(); ++it) {
+ retval.push_back(it->ssrc);
+ }
+ return retval;
+ }
+ // Utility accessor for clients that make the assumption only one ssrc
+ // exists per media.
+ // This will eventually go away.
+ uint32 ssrc() const {
+ if (local_stats.size() > 0) {
+ return local_stats[0].ssrc;
+ } else {
+ return 0;
+ }
+ }
int64 bytes_sent;
int packets_sent;
int packets_lost;
@@ -641,6 +681,35 @@
packets_lost(0),
fraction_lost(0.0) {
}
+ void add_ssrc(const SsrcReceiverInfo& stat) {
+ local_stats.push_back(stat);
+ }
+ // Temporary utility function for call sites that only provide SSRC.
+ // As more info is added into SsrcSenderInfo, this function should go away.
+ void add_ssrc(uint32 ssrc) {
+ SsrcReceiverInfo stat;
+ stat.ssrc = ssrc;
+ add_ssrc(stat);
+ }
+ std::vector<uint32> ssrcs() const {
+ std::vector<uint32> retval;
+ for (std::vector<SsrcReceiverInfo>::const_iterator it = local_stats.begin();
+ it != local_stats.end(); ++it) {
+ retval.push_back(it->ssrc);
+ }
+ return retval;
+ }
+ // Utility accessor for clients that make the assumption only one ssrc
+ // exists per media.
+ // This will eventually go away.
+ uint32 ssrc() const {
+ if (local_stats.size() > 0) {
+ return local_stats[0].ssrc;
+ } else {
+ return 0;
+ }
+ }
+
int64 bytes_rcvd;
int packets_rcvd;
int packets_lost;
@@ -651,8 +720,7 @@
struct VoiceSenderInfo : public MediaSenderInfo {
VoiceSenderInfo()
- : ssrc(0),
- ext_seqnum(0),
+ : ext_seqnum(0),
jitter_ms(0),
audio_level(0),
aec_quality_min(0.0),
@@ -663,7 +731,6 @@
typing_noise_detected(false) {
}
- uint32 ssrc;
int ext_seqnum;
int jitter_ms;
int audio_level;
@@ -677,8 +744,7 @@
struct VoiceReceiverInfo : public MediaReceiverInfo {
VoiceReceiverInfo()
- : ssrc(0),
- ext_seqnum(0),
+ : ext_seqnum(0),
jitter_ms(0),
jitter_buffer_ms(0),
jitter_buffer_preferred_ms(0),
@@ -687,7 +753,6 @@
expand_rate(0) {
}
- uint32 ssrc;
int ext_seqnum;
int jitter_ms;
int jitter_buffer_ms;
@@ -709,10 +774,11 @@
framerate_sent(0),
nominal_bitrate(0),
preferred_bitrate(0),
- adapt_reason(0) {
+ adapt_reason(0),
+ capture_jitter_ms(0),
+ avg_encode_ms(0) {
}
- std::vector<uint32> ssrcs;
std::vector<SsrcGroup> ssrc_groups;
int packets_cached;
int firs_rcvd;
@@ -724,6 +790,8 @@
int nominal_bitrate;
int preferred_bitrate;
int adapt_reason;
+ int capture_jitter_ms;
+ int avg_encode_ms;
};
struct VideoReceiverInfo : public MediaReceiverInfo {
@@ -747,7 +815,6 @@
current_delay_ms(0) {
}
- std::vector<uint32> ssrcs;
std::vector<SsrcGroup> ssrc_groups;
int packets_concealed;
int firs_sent;