Do not add audio bitrate observer if TWCC sending is not supported
Bug: webrtc:8243
Change-Id: Ida076dca72a6894053bdd0884f818ab3eaf5128a
Reviewed-on: https://webrtc-review.googlesource.com/30840
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21149}diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 5a63a6f..ef132cc 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -58,6 +58,7 @@
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../system_wrappers",
+ "../system_wrappers:field_trial_api",
"../voice_engine",
]
}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 2a5628c..1596c96 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -28,6 +28,7 @@
#include "rtc_base/logging.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/timeutils.h"
+#include "system_wrappers/include/field_trial.h"
#include "voice_engine/channel_proxy.h"
#include "voice_engine/include/voe_base.h"
#include "voice_engine/transmit_mixer.h"
@@ -137,6 +138,19 @@
ConfigureStream(this, new_config, false);
}
+AudioSendStream::ExtensionIds AudioSendStream::FindExtensionIds(
+ const std::vector<RtpExtension>& extensions) {
+ ExtensionIds ids;
+ for (const auto& extension : extensions) {
+ if (extension.uri == RtpExtension::kAudioLevelUri) {
+ ids.audio_level = extension.id;
+ } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
+ ids.transport_sequence_number = extension.id;
+ }
+ }
+ return ids;
+}
+
void AudioSendStream::ConfigureStream(
webrtc::internal::AudioSendStream* stream,
const webrtc::AudioSendStream::Config& new_config,
@@ -177,28 +191,8 @@
stream->timed_send_transport_adapter_.get());
}
- // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
- // reserved for padding and MUST NOT be used as a local identifier.
- // So it should be safe to use 0 here to indicate "not configured".
- struct ExtensionIds {
- int audio_level = 0;
- int transport_sequence_number = 0;
- };
-
- auto find_extension_ids = [](const std::vector<RtpExtension>& extensions) {
- ExtensionIds ids;
- for (const auto& extension : extensions) {
- if (extension.uri == RtpExtension::kAudioLevelUri) {
- ids.audio_level = extension.id;
- } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
- ids.transport_sequence_number = extension.id;
- }
- }
- return ids;
- };
-
- const ExtensionIds old_ids = find_extension_ids(old_config.rtp.extensions);
- const ExtensionIds new_ids = find_extension_ids(new_config.rtp.extensions);
+ const ExtensionIds old_ids = FindExtensionIds(old_config.rtp.extensions);
+ const ExtensionIds new_ids = FindExtensionIds(new_config.rtp.extensions);
// Audio level indication
if (first_time || new_ids.audio_level != old_ids.audio_level) {
channel_proxy->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0,
@@ -238,7 +232,10 @@
void AudioSendStream::Start() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) {
+ if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1 &&
+ (FindExtensionIds(config_.rtp.extensions).transport_sequence_number !=
+ 0 ||
+ !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
// Audio BWE is enabled.
transport_->packet_sender()->SetAccountForAudioPackets(true);
ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps);
@@ -599,12 +596,19 @@
// allow us to configure the bitrate observer if the new config has bitrate
// limits set, but would only have us call RemoveBitrateObserver if we were
// previously configured with bitrate limits.
+ int new_transport_seq_num_id =
+ FindExtensionIds(new_config.rtp.extensions).transport_sequence_number;
if (stream->config_.min_bitrate_bps == new_config.min_bitrate_bps &&
- stream->config_.max_bitrate_bps == new_config.max_bitrate_bps) {
+ stream->config_.max_bitrate_bps == new_config.max_bitrate_bps &&
+ (FindExtensionIds(stream->config_.rtp.extensions)
+ .transport_sequence_number == new_transport_seq_num_id ||
+ !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
return;
}
- if (new_config.min_bitrate_bps != -1 && new_config.max_bitrate_bps != -1) {
+ if (new_config.min_bitrate_bps != -1 && new_config.max_bitrate_bps != -1 &&
+ (new_transport_seq_num_id != 0 ||
+ !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
stream->ConfigureBitrateObserver(new_config.min_bitrate_bps,
new_config.max_bitrate_bps);
} else {
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index 08bdddb..1414e39 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -124,6 +124,16 @@
std::unique_ptr<TimedTransport> timed_send_transport_adapter_;
TimeInterval active_lifetime_;
+ // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
+ // reserved for padding and MUST NOT be used as a local identifier.
+ // So it should be safe to use 0 here to indicate "not configured".
+ struct ExtensionIds {
+ int audio_level = 0;
+ int transport_sequence_number = 0;
+ };
+ static ExtensionIds FindExtensionIds(
+ const std::vector<RtpExtension>& extensions);
+
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
};
} // namespace internal
