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(Auto)update libjingle 74825992-> 74851128

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7082 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org
2014-09-05 11:08:01 +00:00
parent 287e9614b3
commit bc6fa1876e
1 changed files with 36 additions and 54 deletions
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90
talk/media/webrtc/webrtcvideoengine.cc
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@@ -2923,39 +2923,10 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
return true;
}
// Trigger SetSendCodec to set correct noise reduction state if the option has
// changed.
bool denoiser_changed = options.video_noise_reduction.IsSet() &&
(options_.video_noise_reduction != options.video_noise_reduction);
bool leaky_bucket_changed = options.video_leaky_bucket.IsSet() &&
(options_.video_leaky_bucket != options.video_leaky_bucket);
bool buffer_latency_changed = options.buffered_mode_latency.IsSet() &&
(options_.buffered_mode_latency != options.buffered_mode_latency);
bool dscp_option_changed = (options_.dscp != options.dscp);
bool suspend_below_min_bitrate_changed =
options.suspend_below_min_bitrate.IsSet() &&
(options_.suspend_below_min_bitrate != options.suspend_below_min_bitrate);
bool conference_mode_turned_off = false;
if (options_.conference_mode.IsSet() && options.conference_mode.IsSet() &&
options_.conference_mode.GetWithDefaultIfUnset(false) &&
!options.conference_mode.GetWithDefaultIfUnset(false)) {
conference_mode_turned_off = true;
}
#ifdef USE_WEBRTC_DEV_BRANCH
bool payload_padding_changed = options.use_payload_padding.IsSet() &&
options_.use_payload_padding != options.use_payload_padding;
#endif
// Save the options, to be interpreted where appropriate.
// Use options_.SetAll() instead of assignment so that unset value in options
// will not overwrite the previous option value.
VideoOptions original = options_;
options_.SetAll(options);
// Set CPU options for all send channels.
@@ -2966,38 +2937,38 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
}
if (send_codec_) {
bool reset_send_codec_needed = denoiser_changed;
webrtc::VideoCodec new_codec = *send_codec_;
bool conference_mode_turned_off = (
original.conference_mode.IsSet() &&
options.conference_mode.IsSet() &&
original.conference_mode.GetWithDefaultIfUnset(false) &&
!options.conference_mode.GetWithDefaultIfUnset(false));
if (conference_mode_turned_off) {
// This is a special case for turning conference mode off.
// Max bitrate should go back to the default maximum value instead
// of the current maximum.
new_codec.maxBitrate = kAutoBandwidth;
reset_send_codec_needed = true;
}
// TODO(pthatcher): Remove this. We don't need 4 ways to set bitrates.
int new_start_bitrate;
if (options.video_start_bitrate.Get(&new_start_bitrate)) {
new_codec.startBitrate = new_start_bitrate;
reset_send_codec_needed = true;
}
LOG(LS_INFO) << "Reset send codec needed is enabled? "
<< reset_send_codec_needed;
if (reset_send_codec_needed) {
if (!SetSendCodec(new_codec)) {
return false;
}
LogSendCodecChange("SetOptions()");
if (!SetSendCodec(new_codec)) {
return false;
}
LogSendCodecChange("SetOptions()");
}
if (leaky_bucket_changed) {
bool enable_leaky_bucket =
options_.video_leaky_bucket.GetWithDefaultIfUnset(true);
bool enable_leaky_bucket;
if (Changed(options.video_leaky_bucket,
original.video_leaky_bucket,
&enable_leaky_bucket)) {
LOG(LS_INFO) << "Leaky bucket is enabled? " << enable_leaky_bucket;
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
@@ -3011,10 +2982,11 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
}
}
}
if (buffer_latency_changed) {
int buffer_latency =
options_.buffered_mode_latency.GetWithDefaultIfUnset(
cricket::kBufferedModeDisabled);
int buffer_latency;
if (Changed(options.buffered_mode_latency,
original.buffered_mode_latency,
&buffer_latency)) {
LOG(LS_INFO) << "Buffer latency is " << buffer_latency;
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
@@ -3033,17 +3005,24 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
}
}
}
if (dscp_option_changed) {
bool dscp_enabled;
if (Changed(options.dscp, original.dscp, &dscp_enabled)) {
rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
if (options_.dscp.GetWithDefaultIfUnset(false))
if (dscp_enabled) {
dscp = kVideoDscpValue;
}
LOG(LS_INFO) << "DSCP is " << dscp;
if (MediaChannel::SetDscp(dscp) != 0) {
LOG(LS_WARNING) << "Failed to set DSCP settings for video channel";
}
}
if (suspend_below_min_bitrate_changed) {
if (options_.suspend_below_min_bitrate.GetWithDefaultIfUnset(false)) {
bool suspend_below_min_bitrate;
if (Changed(options.suspend_below_min_bitrate,
original.suspend_below_min_bitrate,
&suspend_below_min_bitrate)) {
if (suspend_below_min_bitrate) {
LOG(LS_INFO) << "Suspend below min bitrate enabled.";
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
@@ -3054,14 +3033,17 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
LOG(LS_WARNING) << "Cannot disable video suspension once it is enabled";
}
}
#ifdef USE_WEBRTC_DEV_BRANCH
if (payload_padding_changed) {
bool use_payload_pading;
if (Changed(options.use_payload_padding,
original.use_payload_padding,
&use_payload_padding)) {
LOG(LS_INFO) << "Payload-based padding called.";
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
engine()->vie()->rtp()->SetPadWithRedundantPayloads(
it->second->channel_id(),
options_.use_payload_padding.GetWithDefaultIfUnset(false));
it->second->channel_id(), use_payload_padding);
}
}
#endif