Rename AutoMute to SuspendBelowMinBitrate
Changes all instances throughout the WebRTC stack.
BUG=2436
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/video_coding/main/interface/video_coding.h b/webrtc/modules/video_coding/main/interface/video_coding.h
index 5a206e9..9fb98c3 100644
--- a/webrtc/modules/video_coding/main/interface/video_coding.h
+++ b/webrtc/modules/video_coding/main/interface/video_coding.h
@@ -592,17 +592,14 @@
// Disables recording of debugging information.
virtual int StopDebugRecording() = 0;
- // Enables AutoMuter to turn off video when the rate drops below
+ // Lets the sender suspend video when the rate drops below
// |threshold_bps|, and turns back on when the rate goes back up above
// |threshold_bps| + |window_bps|.
- virtual void EnableAutoMuting() = 0;
+ virtual void SuspendBelowMinBitrate() = 0;
- // Disables AutoMuter.
- virtual void DisableAutoMuting() = 0;
-
- // Returns true if AutoMuter is engaged and the video has been muted due to
- // bandwidth limitations; otherwise false.
- virtual bool VideoMuted() const = 0;
+ // Returns true if SuspendBelowMinBitrate is engaged and the video has been
+ // suspended due to bandwidth limitations; otherwise false.
+ virtual bool VideoSuspended() const = 0;
};
} // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/media_optimization.cc b/webrtc/modules/video_coding/main/source/media_optimization.cc
index 815be36..37dff6c 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization.cc
+++ b/webrtc/modules/video_coding/main/source/media_optimization.cc
@@ -47,10 +47,10 @@
last_qm_update_time_(0),
last_change_time_(0),
num_layers_(0),
- muting_enabled_(false),
- video_muted_(false),
- muter_threshold_bps_(0),
- muter_window_bps_(0) {
+ suspension_enabled_(false),
+ video_suspended_(false),
+ suspension_threshold_bps_(0),
+ suspension_window_bps_(0) {
memset(send_statistics_, 0, sizeof(send_statistics_));
memset(incoming_frame_times_, -1, sizeof(incoming_frame_times_));
}
@@ -193,7 +193,7 @@
content_->ResetShortTermAvgData();
}
- CheckAutoMuteConditions();
+ CheckSuspendConditions();
return target_bit_rate_;
}
@@ -351,7 +351,7 @@
bool MediaOptimization::DropFrame() {
// Leak appropriate number of bytes.
frame_dropper_->Leak((uint32_t)(InputFrameRate() + 0.5f));
- if (video_muted_) {
+ if (video_suspended_) {
return true; // Drop all frames when muted.
}
return frame_dropper_->DropFrame();
@@ -418,17 +418,13 @@
return VCM_OK;
}
-void MediaOptimization::EnableAutoMuting(int threshold_bps, int window_bps) {
+void MediaOptimization::SuspendBelowMinBitrate(int threshold_bps,
+ int window_bps) {
assert(threshold_bps > 0 && window_bps >= 0);
- muter_threshold_bps_ = threshold_bps;
- muter_window_bps_ = window_bps;
- muting_enabled_ = true;
- video_muted_ = false;
-}
-
-void MediaOptimization::DisableAutoMuting() {
- muting_enabled_ = false;
- video_muted_ = false;
+ suspension_threshold_bps_ = threshold_bps;
+ suspension_window_bps_ = window_bps;
+ suspension_enabled_ = true;
+ video_suspended_ = false;
}
// Private methods below this line.
@@ -605,19 +601,20 @@
}
}
-void MediaOptimization::CheckAutoMuteConditions() {
- // Check conditions for AutoMute. |target_bit_rate_| is in bps.
- if (muting_enabled_) {
- if (!video_muted_) {
+void MediaOptimization::CheckSuspendConditions() {
+ // Check conditions for SuspendBelowMinBitrate. |target_bit_rate_| is in bps.
+ if (suspension_enabled_) {
+ if (!video_suspended_) {
// Check if we just went below the threshold.
- if (target_bit_rate_ < muter_threshold_bps_) {
- video_muted_ = true;
+ if (target_bit_rate_ < suspension_threshold_bps_) {
+ video_suspended_ = true;
}
} else {
- // Video is already muted. Check if we just went over the threshold
+ // Video is already suspended. Check if we just went over the threshold
// with a margin.
- if (target_bit_rate_ > muter_threshold_bps_ + muter_window_bps_) {
- video_muted_ = false;
+ if (target_bit_rate_ >
+ suspension_threshold_bps_ + suspension_window_bps_) {
+ video_suspended_ = false;
}
}
}
diff --git a/webrtc/modules/video_coding/main/source/media_optimization.h b/webrtc/modules/video_coding/main/source/media_optimization.h
index ca383bf..cde28d2 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization.h
+++ b/webrtc/modules/video_coding/main/source/media_optimization.h
@@ -120,18 +120,15 @@
// Computes new Quality Mode.
int32_t SelectQuality();
- // Enables AutoMuter to turn off video when the rate drops below
+ // Lets the sender suspend video when the rate drops below
// |threshold_bps|, and turns back on when the rate goes back up above
// |threshold_bps| + |window_bps|.
- void EnableAutoMuting(int threshold_bps, int window_bps);
-
- // Disables AutoMuter.
- void DisableAutoMuting();
+ void SuspendBelowMinBitrate(int threshold_bps, int window_bps);
// Accessors and mutators.
int32_t max_bit_rate() const { return max_bit_rate_; }
void set_max_payload_size(int32_t mtu) { max_payload_size_ = mtu; }
- bool video_muted() const { return video_muted_; }
+ bool video_suspended() const { return video_suspended_; }
private:
typedef std::list<EncodedFrameSample> FrameSampleList;
@@ -161,10 +158,10 @@
void ProcessIncomingFrameRate(int64_t now);
- // Checks conditions for AutoMute. The method compares |target_bit_rate_|
- // with the threshold values for AutoMute, and changes the state of
- // |video_muted_| accordingly.
- void CheckAutoMuteConditions();
+ // Checks conditions for suspending the video. The method compares
+ // |target_bit_rate_| with the threshold values for suspension, and changes
+ // the state of |video_suspended_| accordingly.
+ void CheckSuspendConditions();
int32_t id_;
Clock* clock_;
@@ -195,10 +192,10 @@
int64_t last_qm_update_time_;
int64_t last_change_time_; // Content/user triggered.
int num_layers_;
- bool muting_enabled_;
- bool video_muted_;
- int muter_threshold_bps_;
- int muter_window_bps_;
+ bool suspension_enabled_;
+ bool video_suspended_;
+ int suspension_threshold_bps_;
+ int suspension_window_bps_;
}; // End of MediaOptimization class declaration.
} // namespace media_optimization
diff --git a/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc b/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc
index fc4fd75..1425dad 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc
@@ -55,15 +55,15 @@
TEST_F(TestMediaOptimization, VerifyMuting) {
- // Enable video muter with these limits.
- // Mute the video when the rate is below 50 kbps and unmute when it gets above
- // 50 + 10 kbps again.
+ // Enable video suspension with these limits.
+ // Suspend the video when the rate is below 50 kbps and resume when it gets
+ // above 50 + 10 kbps again.
const int kThresholdBps = 50000;
const int kWindowBps = 10000;
- media_opt_.EnableAutoMuting(kThresholdBps, kWindowBps);
+ media_opt_.SuspendBelowMinBitrate(kThresholdBps, kWindowBps);
- // The video should not be muted from the start.
- EXPECT_FALSE(media_opt_.video_muted());
+ // The video should not be suspended from the start.
+ EXPECT_FALSE(media_opt_.video_suspended());
int target_bitrate_kbps = 100;
media_opt_.SetTargetRates(target_bitrate_kbps * 1000,
@@ -81,7 +81,7 @@
// Expect the muter to engage immediately and stay muted.
// Test during 2 seconds.
for (int time = 0; time < 2000; time += frame_time_ms_) {
- EXPECT_TRUE(media_opt_.video_muted());
+ EXPECT_TRUE(media_opt_.video_suspended());
ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, true));
}
@@ -93,7 +93,7 @@
// Expect the muter to stay muted.
// Test during 2 seconds.
for (int time = 0; time < 2000; time += frame_time_ms_) {
- EXPECT_TRUE(media_opt_.video_muted());
+ EXPECT_TRUE(media_opt_.video_suspended());
ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, true));
}
@@ -104,7 +104,7 @@
// Expect the muter to disengage immediately.
// Test during 2 seconds.
for (int time = 0; time < 2000; time += frame_time_ms_) {
- EXPECT_FALSE(media_opt_.video_muted());
+ EXPECT_FALSE(media_opt_.video_suspended());
ASSERT_NO_FATAL_FAILURE(
AddFrameAndAdvanceTime((kThresholdBps + kWindowBps) / 1000, false));
}
diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl.cc b/webrtc/modules/video_coding/main/source/video_coding_impl.cc
index 9524984..8282edc 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_impl.cc
+++ b/webrtc/modules/video_coding/main/source/video_coding_impl.cc
@@ -197,16 +197,12 @@
return sender_->StopDebugRecording();
}
- virtual void EnableAutoMuting() {
- return sender_->EnableAutoMuting();
+ virtual void SuspendBelowMinBitrate() {
+ return sender_->SuspendBelowMinBitrate();
}
- virtual void DisableAutoMuting() {
- return sender_->DisableAutoMuting();
- }
-
- virtual bool VideoMuted() const {
- return sender_->VideoMuted();
+ virtual bool VideoSuspended() const {
+ return sender_->VideoSuspended();
}
virtual int32_t InitializeReceiver() OVERRIDE {
diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl.h b/webrtc/modules/video_coding/main/source/video_coding_impl.h
index f9a7973..c1d02d9 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_impl.h
+++ b/webrtc/modules/video_coding/main/source/video_coding_impl.h
@@ -95,9 +95,8 @@
int StartDebugRecording(const char* file_name_utf8);
int StopDebugRecording();
- void EnableAutoMuting();
- void DisableAutoMuting();
- bool VideoMuted() const;
+ void SuspendBelowMinBitrate();
+ bool VideoSuspended() const;
int32_t TimeUntilNextProcess();
int32_t Process();
diff --git a/webrtc/modules/video_coding/main/source/video_sender.cc b/webrtc/modules/video_coding/main/source/video_sender.cc
index 4cbcc89..5a9a563 100644
--- a/webrtc/modules/video_coding/main/source/video_sender.cc
+++ b/webrtc/modules/video_coding/main/source/video_sender.cc
@@ -422,11 +422,11 @@
return VCM_OK;
}
-void VideoSender::EnableAutoMuting() {
+void VideoSender::SuspendBelowMinBitrate() {
CriticalSectionScoped cs(_sendCritSect);
VideoCodec current_send_codec;
if (SendCodec(¤t_send_codec) != 0) {
- assert(false); // Must set a send codec before enabling auto-mute.
+ assert(false); // Must set a send codec before SuspendBelowMinBitrate.
return;
}
int threshold_bps;
@@ -438,17 +438,12 @@
// Set the hysteresis window to be at 10% of the threshold, but at least
// 10 kbps.
int window_bps = std::max(threshold_bps / 10, 10000);
- _mediaOpt.EnableAutoMuting(threshold_bps, window_bps);
+ _mediaOpt.SuspendBelowMinBitrate(threshold_bps, window_bps);
}
-void VideoSender::DisableAutoMuting() {
+bool VideoSender::VideoSuspended() const {
CriticalSectionScoped cs(_sendCritSect);
- _mediaOpt.DisableAutoMuting();
-}
-
-bool VideoSender::VideoMuted() const {
- CriticalSectionScoped cs(_sendCritSect);
- return _mediaOpt.video_muted();
+ return _mediaOpt.video_suspended();
}
} // namespace vcm
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 5ae1e94..e1a60ed 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -198,8 +198,8 @@
image_process_->RegisterPreEncodeCallback(channel_,
config_.pre_encode_callback);
- if (config.auto_mute) {
- codec_->EnableAutoMuting(channel_);
+ if (config.suspend_below_min_bitrate) {
+ codec_->SuspendBelowMinBitrate(channel_);
}
}
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 2cb94ab..7e12713 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -550,13 +550,13 @@
// The test will go through a number of phases.
// 1. Start sending packets.
// 2. As soon as the RTP stream has been detected, signal a low REMB value to
-// activate the auto muter.
-// 3. Wait until |kMuteTimeFrames| have been captured without seeing any RTP
+// suspend the stream.
+// 3. Wait until |kSuspendTimeFrames| have been captured without seeing any RTP
// packets.
-// 4. Signal a high REMB and the wait for the RTP stream to start again.
+// 4. Signal a high REMB and then wait for the RTP stream to start again.
// When the stream is detected again, the test ends.
-TEST_F(VideoSendStreamTest, AutoMute) {
- static const int kMuteTimeFrames = 60; // Mute for 2 seconds @ 30 fps.
+TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
+ static const int kSuspendTimeFrames = 60; // Suspend for 2 seconds @ 30 fps.
class RembObserver : public test::RtpRtcpObserver, public I420FrameCallback {
public:
@@ -564,10 +564,10 @@
: RtpRtcpObserver(30 * 1000), // Timeout after 30 seconds.
transport_adapter_(&transport_),
clock_(Clock::GetRealTimeClock()),
- test_state_(kBeforeMute),
+ test_state_(kBeforeSuspend),
rtp_count_(0),
last_sequence_number_(0),
- mute_frame_count_(0),
+ suspended_frame_count_(0),
low_remb_bps_(0),
high_remb_bps_(0),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
@@ -591,12 +591,12 @@
EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header));
last_sequence_number_ = header.sequenceNumber;
- if (test_state_ == kBeforeMute) {
- // The stream has started. Try to mute it.
+ if (test_state_ == kBeforeSuspend) {
+ // The stream has started. Try to suspend it.
SendRtcpFeedback(low_remb_bps_);
- test_state_ = kDuringMute;
- } else if (test_state_ == kDuringMute) {
- mute_frame_count_ = 0;
+ test_state_ = kDuringSuspend;
+ } else if (test_state_ == kDuringSuspend) {
+ suspended_frame_count_ = 0;
} else if (test_state_ == kWaitingForPacket) {
observation_complete_->Set();
}
@@ -607,7 +607,8 @@
// This method implements the I420FrameCallback.
void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
CriticalSectionScoped lock(crit_sect_.get());
- if (test_state_ == kDuringMute && ++mute_frame_count_ > kMuteTimeFrames) {
+ if (test_state_ == kDuringSuspend &&
+ ++suspended_frame_count_ > kSuspendTimeFrames) {
SendRtcpFeedback(high_remb_bps_);
test_state_ = kWaitingForPacket;
}
@@ -621,10 +622,10 @@
private:
enum TestState {
- kBeforeMute,
- kDuringMute,
+ kBeforeSuspend,
+ kDuringSuspend,
kWaitingForPacket,
- kAfterMute
+ kAfterSuspend
};
virtual void SendRtcpFeedback(int remb_value) {
@@ -649,7 +650,7 @@
TestState test_state_;
int rtp_count_;
int last_sequence_number_;
- int mute_frame_count_;
+ int suspended_frame_count_;
int low_remb_bps_;
int high_remb_bps_;
scoped_ptr<CriticalSectionWrapper> crit_sect_;
@@ -662,7 +663,7 @@
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.nack.rtp_history_ms = 1000;
send_config.pre_encode_callback = &observer;
- send_config.auto_mute = true;
+ send_config.suspend_below_min_bitrate = true;
unsigned int min_bitrate_bps =
send_config.codec.simulcastStream[0].minBitrate * 1000;
observer.set_low_remb_bps(min_bitrate_bps - 10000);
diff --git a/webrtc/video_engine/include/vie_codec.h b/webrtc/video_engine/include/vie_codec.h
index f470438..d5b5602 100644
--- a/webrtc/video_engine/include/vie_codec.h
+++ b/webrtc/video_engine/include/vie_codec.h
@@ -36,10 +36,10 @@
const unsigned int framerate,
const unsigned int bitrate) = 0;
- // This method is called whenever the state of the AutoMuter changes, i.e.,
- // when |is_muted| toggles.
+ // This method is called whenever the state of the SuspendBelowMinBitrate
+ // changes, i.e., when |is_suspended| toggles.
// TODO(hlundin): Remove the default implementation when possible.
- virtual void VideoAutoMuted(int video_channel, bool is_muted) {}
+ virtual void VideoSuspended(int video_channel, bool is_suspended) {}
protected:
virtual ~ViEEncoderObserver() {}
@@ -193,12 +193,12 @@
// Disables recording of debugging information.
virtual int StopDebugRecording(int video_channel) = 0;
- // Enables AutoMuter to turn off video when the rate drops below
+ // Lets the sender suspend video when the rate drops below
// |threshold_bps|, and turns back on when the rate goes back up above
// |threshold_bps| + |window_bps|.
// This is under development; not tested.
// TODO(hlundin): Remove the default implementation when possible.
- virtual void EnableAutoMuting(int video_channel) {}
+ virtual void SuspendBelowMinBitrate(int video_channel) {}
protected:
ViECodec() {}
diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
index 3fa2bc4..b3ca926 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
@@ -44,7 +44,7 @@
unsigned int last_outgoing_bitrate_;
unsigned int last_incoming_framerate_;
unsigned int last_incoming_bitrate_;
- unsigned int video_auto_muted_called_;
+ unsigned int video_suspended_called_;
webrtc::VideoCodec incoming_codec_;
@@ -60,7 +60,7 @@
last_outgoing_bitrate_(0),
last_incoming_framerate_(0),
last_incoming_bitrate_(0),
- video_auto_muted_called_(0) {
+ video_suspended_called_(0) {
memset(&incoming_codec_, 0, sizeof(incoming_codec_));
}
virtual void IncomingCodecChanged(const int video_channel,
@@ -100,8 +100,8 @@
last_outgoing_bitrate_ += bitrate;
}
- virtual void VideoAutoMuted(int video_channel, bool is_muted) {
- video_auto_muted_called_++;
+ virtual void VideoSuspended(int video_channel, bool is_suspended) OVERRIDE {
+ video_suspended_called_++;
}
virtual void RequestNewKeyFrame(const int video_channel) {
diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
index 8d7613d..5d3fdc4 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
@@ -73,8 +73,8 @@
<< " BR: " << bitrate << std::endl;
}
- virtual void VideoAutoMuted(int video_channel, bool is_muted) {
- std::cout << "VideoAutoMuted: " << is_muted << std::endl;
+ virtual void VideoSuspended(int video_channel, bool is_suspended) OVERRIDE {
+ std::cout << "VideoSuspended: " << is_suspended << std::endl;
}
};
diff --git a/webrtc/video_engine/vie_codec_impl.cc b/webrtc/video_engine/vie_codec_impl.cc
index 4d927b5..94e4d7c 100644
--- a/webrtc/video_engine/vie_codec_impl.cc
+++ b/webrtc/video_engine/vie_codec_impl.cc
@@ -715,7 +715,7 @@
return vie_encoder->StopDebugRecording();
}
-void ViECodecImpl::EnableAutoMuting(int video_channel) {
+void ViECodecImpl::SuspendBelowMinBitrate(int video_channel) {
ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
ViEEncoder* vie_encoder = cs.Encoder(video_channel);
if (!vie_encoder) {
@@ -724,7 +724,7 @@
"%s: No encoder %d", __FUNCTION__, video_channel);
return;
}
- return vie_encoder->EnableAutoMuting();
+ return vie_encoder->SuspendBelowMinBitrate();
}
bool ViECodecImpl::CodecValid(const VideoCodec& video_codec) {
diff --git a/webrtc/video_engine/vie_codec_impl.h b/webrtc/video_engine/vie_codec_impl.h
index 8bcac41..372ffc9 100644
--- a/webrtc/video_engine/vie_codec_impl.h
+++ b/webrtc/video_engine/vie_codec_impl.h
@@ -70,7 +70,7 @@
virtual int StartDebugRecording(int video_channel,
const char* file_name_utf8);
virtual int StopDebugRecording(int video_channel);
- virtual void EnableAutoMuting(int video_channel);
+ virtual void SuspendBelowMinBitrate(int video_channel);
protected:
explicit ViECodecImpl(ViESharedData* shared_data);
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 889842e..65a9e8b 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -163,7 +163,7 @@
has_received_rpsi_(false),
picture_id_rpsi_(0),
qm_callback_(NULL),
- video_auto_muted_(false),
+ video_suspended_(false),
pre_encode_callback_(NULL) {
WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo,
ViEId(engine_id, channel_id),
@@ -1050,7 +1050,7 @@
__FUNCTION__, bitrate_bps, fraction_lost, round_trip_time_ms);
vcm_.SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
- bool video_is_muted = vcm_.VideoMuted();
+ bool video_is_suspended = vcm_.VideoSuspended();
int bitrate_kbps = bitrate_bps / 1000;
VideoCodec send_codec;
if (vcm_.SendCodec(&send_codec) != 0) {
@@ -1087,10 +1087,10 @@
pad_up_to_bitrate_kbps += stream_configs[i].targetBitrate;
}
}
- if (video_is_muted || send_codec.numberOfSimulcastStreams > 1) {
+ if (video_is_suspended || send_codec.numberOfSimulcastStreams > 1) {
pad_up_to_bitrate_kbps = std::min(bitrate_kbps, pad_up_to_bitrate_kbps);
} else {
- // Disable padding if only sending one stream and video isn't muted.
+ // Disable padding if only sending one stream and video isn't suspended.
pad_up_to_bitrate_kbps = 0;
}
@@ -1107,15 +1107,15 @@
max_padding_bitrate_kbps,
pad_up_to_bitrate_kbps);
default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
- if (video_is_muted != video_auto_muted_) {
+ if (video_is_suspended != video_suspended_) {
// State changed now. Send callback to inform about that.
- video_auto_muted_ = video_is_muted;
+ video_suspended_ = video_is_suspended;
if (codec_observer_) {
WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
ViEId(engine_id_, channel_id_),
- "%s: video_auto_muted_ changed to %i",
- __FUNCTION__, video_auto_muted_);
- codec_observer_->VideoAutoMuted(channel_id_, video_auto_muted_);
+ "%s: video_suspended_ changed to %i",
+ __FUNCTION__, video_suspended_);
+ codec_observer_->VideoSuspended(channel_id_, video_suspended_);
}
}
}
@@ -1159,8 +1159,8 @@
return vcm_.StopDebugRecording();
}
-void ViEEncoder::EnableAutoMuting() {
- vcm_.EnableAutoMuting();
+void ViEEncoder::SuspendBelowMinBitrate() {
+ vcm_.SuspendBelowMinBitrate();
bitrate_controller_->EnforceMinBitrate(false);
}
diff --git a/webrtc/video_engine/vie_encoder.h b/webrtc/video_engine/vie_encoder.h
index f80699c..73f84af 100644
--- a/webrtc/video_engine/vie_encoder.h
+++ b/webrtc/video_engine/vie_encoder.h
@@ -163,10 +163,10 @@
// Disables recording of debugging information.
virtual int StopDebugRecording();
- // Enables AutoMuter to turn off video when the rate drops below
+ // Lets the sender suspend video when the rate drops below
// |threshold_bps|, and turns back on when the rate goes back up above
// |threshold_bps| + |window_bps|.
- virtual void EnableAutoMuting();
+ virtual void SuspendBelowMinBitrate();
// New-style callback, used by VideoSendStream.
void RegisterPreEncodeCallback(I420FrameCallback* pre_encode_callback);
@@ -226,7 +226,7 @@
// Quality modes callback
QMVideoSettingsCallback* qm_callback_;
- bool video_auto_muted_;
+ bool video_suspended_;
I420FrameCallback* pre_encode_callback_;
};
diff --git a/webrtc/video_send_stream.h b/webrtc/video_send_stream.h
index 2df282b..ee5826d 100644
--- a/webrtc/video_send_stream.h
+++ b/webrtc/video_send_stream.h
@@ -79,7 +79,7 @@
target_delay_ms(0),
pacing(false),
stats_callback(NULL),
- auto_mute(false) {}
+ suspend_below_min_bitrate(false) {}
VideoCodec codec;
static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
@@ -142,10 +142,10 @@
// Callback for periodically receiving send stats.
StatsCallback* stats_callback;
- // True if video should be muted when video goes under the minimum video
- // bitrate. Unless muted, video will be sent at a bitrate higher than
- // estimated available.
- bool auto_mute;
+ // True if the stream should be suspended when the available bitrate fall
+ // below the minimum configured bitrate. If this variable is false, the
+ // stream may send at a rate higher than the estimated available bitrate.
+ bool suspend_below_min_bitrate;
};
// Gets interface used to insert captured frames. Valid as long as the
