Fix two invalid DCHECKs related to audio BWE.
These are invalid since:
- An allocated bitrate of 0 means that the stream should be disabled. Changing the behavior to send audio at min bitrate even though the allocator asks for the stream to be disabled.
- It should be OK to set a min bitrate equal to the start bitrate.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2806163003
Cr-Commit-Position: refs/heads/master@{#17613}
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 078aea6..7f0d825 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -246,6 +246,12 @@
uint8_t fraction_loss,
int64_t rtt,
int64_t probing_interval_ms) {
+ // A send stream may be allocated a bitrate of zero if the allocator decides
+ // to disable it. For now we ignore this decision and keep sending on min
+ // bitrate.
+ if (bitrate_bps == 0) {
+ bitrate_bps = config_.min_bitrate_bps;
+ }
RTC_DCHECK_GE(bitrate_bps,
static_cast<uint32_t>(config_.min_bitrate_bps));
// The bitrate allocator might allocate an higher than max configured bitrate
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 5fcacd1..e381183 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -374,7 +374,7 @@
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK(config.event_log != nullptr);
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
- RTC_DCHECK_GT(config.bitrate_config.start_bitrate_bps,
+ RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
config.bitrate_config.min_bitrate_bps);
if (config.bitrate_config.max_bitrate_bps != -1) {
RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
