Routing BitrateAllocationUpdate to audio codec.
This will be used in a later CL to use the link capacity field in the
update to control the Opus encoder.
Bug: webrtc:9718
Change-Id: If2ad16a8f4656e8cdf10c33f5fb060ef7ca5caba
Reviewed-on: https://webrtc-review.googlesource.com/c/111510
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25761}diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index e8b7b32..31fe25f 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -111,7 +111,7 @@
void StopSend() override;
// Codecs
- void SetBitrate(int bitrate_bps, int64_t probing_interval_ms) override;
+ void OnBitrateAllocation(BitrateAllocationUpdate update) override;
int GetBitrate() const override;
// Network
@@ -898,7 +898,7 @@
audio_coding_->ModifyEncoder(modifier);
}
-void ChannelSend::SetBitrate(int bitrate_bps, int64_t probing_interval_ms) {
+void ChannelSend::OnBitrateAllocation(BitrateAllocationUpdate update) {
// This method can be called on the worker thread, module process thread
// or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
// TODO(solenberg): Figure out a good way to check this or enforce calling
@@ -909,11 +909,11 @@
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
if (*encoder) {
- (*encoder)->OnReceivedUplinkBandwidth(bitrate_bps, probing_interval_ms);
+ (*encoder)->OnReceivedUplinkAllocation(update);
}
});
- retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
- configured_bitrate_bps_ = bitrate_bps;
+ retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps());
+ configured_bitrate_bps_ = update.target_bitrate.bps();
}
int ChannelSend::GetBitrate() const {