Index: webrtc/voice_engine/channel.cc |
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
index 100832de3adeb2dc986cd255dc271a109ea6bbb7..49ed483148e22564c630171dfe0d7ddd1e917d10 100644 |
--- a/webrtc/voice_engine/channel.cc |
+++ b/webrtc/voice_engine/channel.cc |
@@ -924,10 +924,7 @@ Channel::Channel(int32_t channelId, |
rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), |
retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), |
kMaxRetransmissionWindowMs)), |
- decoder_factory_(config.acm_config.decoder_factory), |
- // Bitrate smoother can be initialized with arbitrary time constant |
- // (0 used here). The actual time constant will be set in SetBitRate. |
- bitrate_smoother_(0, Clock::GetRealTimeClock()) { |
+ decoder_factory_(config.acm_config.decoder_factory) { |
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), |
"Channel::Channel() - ctor"); |
AudioCodingModule::Config acm_config(config.acm_config); |
@@ -1332,28 +1329,10 @@ void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) { |
"Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps); |
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
if (*encoder) |
minyue-webrtc
2016/12/22 14:51:27
{
...
}
michaelt
2016/12/22 15:10:10
Done.
|
- (*encoder)->OnReceivedTargetAudioBitrate(bitrate_bps); |
+ (*encoder)->OnReceivedTargetAudioBitrate( |
+ bitrate_bps, rtc::Optional<int64_t>(probing_interval_ms)); |
}); |
retransmission_rate_limiter_->SetMaxRate(bitrate_bps); |
- |
- // We give smoothed bitrate allocation to audio network adaptor as |
- // the uplink bandwidth. |
- // The probing spikes should not affect the bitrate smoother more than 25%. |
- // To simplify the calculations we use a step response as input signal. |
- // The step response of an exponential filter is |
- // u(t) = 1 - e^(-t / time_constant). |
- // In order to limit the affect of a BWE spike within 25% of its value before |
- // the next probing, we would choose a time constant that fulfills |
- // 1 - e^(-probing_interval_ms / time_constant) < 0.25 |
- // Then 4 * probing_interval_ms is a good choice. |
- bitrate_smoother_.SetTimeConstantMs(probing_interval_ms * 4); |
- bitrate_smoother_.AddSample(bitrate_bps); |
- audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
- if (*encoder) { |
- (*encoder)->OnReceivedUplinkBandwidth( |
- static_cast<int>(*bitrate_smoother_.GetAverage())); |
- } |
- }); |
} |
void Channel::OnIncomingFractionLoss(int fraction_lost) { |