| Index: webrtc/audio/audio_send_stream.cc
|
| diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
|
| index ad6366bc8095deebe830b33c9115cf77afe83adf..7e8e426ed9a2eeb42cc8d57ffcdc38c8d7960e88 100644
|
| --- a/webrtc/audio/audio_send_stream.cc
|
| +++ b/webrtc/audio/audio_send_stream.cc
|
| @@ -101,12 +101,12 @@ AudioSendStream::~AudioSendStream() {
|
|
|
| void AudioSendStream::Start() {
|
| RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
| - if (config_.min_bitrate_kbps != -1 && config_.max_bitrate_kbps != -1) {
|
| - RTC_DCHECK_GE(config_.max_bitrate_kbps, config_.min_bitrate_kbps);
|
| + if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) {
|
| + RTC_DCHECK_GE(config_.max_bitrate_bps, config_.min_bitrate_bps);
|
| rtc::Event thread_sync_event(false /* manual_reset */, false);
|
| worker_queue_->PostTask([this, &thread_sync_event] {
|
| - bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000,
|
| - config_.max_bitrate_kbps * 1000, 0, true);
|
| + bitrate_allocator_->AddObserver(this, config_.min_bitrate_bps,
|
| + config_.max_bitrate_bps, 0, true);
|
| thread_sync_event.Set();
|
| });
|
| thread_sync_event.Wait(rtc::Event::kForever);
|
| @@ -249,10 +249,10 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
|
| uint8_t fraction_loss,
|
| int64_t rtt) {
|
| RTC_DCHECK_GE(bitrate_bps,
|
| - static_cast<uint32_t>(config_.min_bitrate_kbps * 1000));
|
| + static_cast<uint32_t>(config_.min_bitrate_bps));
|
| // The bitrate allocator might allocate an higher than max configured bitrate
|
| // if there is room, to allow for, as example, extra FEC. Ignore that for now.
|
| - const uint32_t max_bitrate_bps = config_.max_bitrate_kbps * 1000;
|
| + const uint32_t max_bitrate_bps = config_.max_bitrate_bps;
|
| if (bitrate_bps > max_bitrate_bps)
|
| bitrate_bps = max_bitrate_bps;
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|
|
|
|