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Do not add audio bitrate observer if TWCC sending is not supported

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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}
This commit is contained in:
Alex Narest
2017-12-07 20:54:55 +01:00
committed by Commit Bot
parent 6de95f06d0
commit cedd351e73
3 changed files with 40 additions and 25 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
audio/BUILD.gn
View File

@ -58,6 +58,7 @@ rtc_static_library("audio") {
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../voice_engine",
]
}

54
audio/audio_send_stream.cc
View File

@ -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 @@ void AudioSendStream::Reconfigure(
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 @@ void AudioSendStream::ConfigureStream(
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::ConfigureStream(
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 @@ void AudioSendStream::ReconfigureBitrateObserver(
// 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 {

10
audio/audio_send_stream.h
View File

@ -124,6 +124,16 @@ class AudioSendStream final : public webrtc::AudioSendStream,
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