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Trigger audio bitrate allocation update on overhead change.

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This prepares for adding correct overhead calculation to audio bitrate
allocation.

Bug: webrtc:10286
Change-Id: I4669203269396195f7f2ad412ae8470d091e8930
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125090
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27001}
This commit is contained in:
Sebastian Jansson 2019-03-01 15:57:55 +01:00 committed by Commit Bot
parent ee5ccbc57f
commit 8672cac32b
2 changed files with 54 additions and 32 deletions
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76
audio/audio_send_stream.cc
View File

@ -179,6 +179,11 @@ AudioSendStream::~AudioSendStream() {
rtp_transport_->DeRegisterPacketFeedbackObserver(this);
channel_send_->ResetSenderCongestionControlObjects();
}
// Blocking call to synchronize state with worker queue to ensure that there
// are no pending tasks left that keeps references to audio.
rtc::Event thread_sync_event;
worker_queue_->PostTask([&] { thread_sync_event.Set(); });
thread_sync_event.Wait(rtc::Event::kForever);
}
const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
@ -310,7 +315,7 @@ void AudioSendStream::ConfigureStream(
}
void AudioSendStream::Start() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (sending_) {
return;
}
@ -320,8 +325,13 @@ void AudioSendStream::Start() {
TransportSeqNumId(config_))) {
rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps,
config_.bitrate_priority);
rtc::Event thread_sync_event;
worker_queue_->PostTask([&] {
RTC_DCHECK_RUN_ON(worker_queue_);
ConfigureBitrateObserver();
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
} else {
rtp_rtcp_module_->SetAsPartOfAllocation(false);
}
@ -505,6 +515,15 @@ void AudioSendStream::UpdateOverheadForEncoder() {
channel_send_->CallEncoder([&](AudioEncoder* encoder) {
encoder->OnReceivedOverhead(overhead_per_packet_bytes);
});
worker_queue_->PostTask([this, overhead_per_packet_bytes] {
RTC_DCHECK_RUN_ON(worker_queue_);
if (total_packet_overhead_bytes_ != overhead_per_packet_bytes) {
total_packet_overhead_bytes_ = overhead_per_packet_bytes;
if (registered_with_allocator_) {
ConfigureBitrateObserver();
}
}
});
}
size_t AudioSendStream::TestOnlyGetPerPacketOverheadBytes() const {
@ -733,6 +752,7 @@ void AudioSendStream::ReconfigureCNG(AudioSendStream* stream,
void AudioSendStream::ReconfigureBitrateObserver(
AudioSendStream* stream,
const webrtc::AudioSendStream::Config& new_config) {
RTC_DCHECK_RUN_ON(&stream->worker_thread_checker_);
// Since the Config's default is for both of these to be -1, this test will
// 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
@ -749,9 +769,20 @@ void AudioSendStream::ReconfigureBitrateObserver(
new_config.min_bitrate_bps, new_config.max_bitrate_bps,
new_config.has_dscp, TransportSeqNumId(new_config))) {
stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
stream->ConfigureBitrateObserver(new_config.min_bitrate_bps,
new_config.max_bitrate_bps,
new_config.bitrate_priority);
rtc::Event thread_sync_event;
stream->worker_queue_->PostTask([&] {
RTC_DCHECK_RUN_ON(stream->worker_queue_);
stream->registered_with_allocator_ = true;
// We may get a callback immediately as the observer is registered, so
// make
// sure the bitrate limits in config_ are up-to-date.
stream->config_.min_bitrate_bps = new_config.min_bitrate_bps;
stream->config_.max_bitrate_bps = new_config.max_bitrate_bps;
stream->config_.bitrate_priority = new_config.bitrate_priority;
stream->ConfigureBitrateObserver();
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
stream->rtp_rtcp_module_->SetAsPartOfAllocation(true);
} else {
stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(false);
@ -760,34 +791,23 @@ void AudioSendStream::ReconfigureBitrateObserver(
}
}
void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps,
int max_bitrate_bps,
double bitrate_priority) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(max_bitrate_bps, min_bitrate_bps);
rtc::Event thread_sync_event;
worker_queue_->PostTask([&] {
// We may get a callback immediately as the observer is registered, so make
// sure the bitrate limits in config_ are up-to-date.
config_.min_bitrate_bps = min_bitrate_bps;
config_.max_bitrate_bps = max_bitrate_bps;
config_.bitrate_priority = bitrate_priority;
// This either updates the current observer or adds a new observer.
bitrate_allocator_->AddObserver(
this, MediaStreamAllocationConfig{
static_cast<uint32_t>(min_bitrate_bps),
static_cast<uint32_t>(max_bitrate_bps), 0,
allocation_settings_.DefaultPriorityBitrate().bps(), true,
config_.track_id, bitrate_priority});
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
void AudioSendStream::ConfigureBitrateObserver() {
// This either updates the current observer or adds a new observer.
// TODO(srte): Add overhead compensation here.
bitrate_allocator_->AddObserver(
this, MediaStreamAllocationConfig{
static_cast<uint32_t>(config_.min_bitrate_bps),
static_cast<uint32_t>(config_.max_bitrate_bps), 0,
allocation_settings_.DefaultPriorityBitrate().bps(), true,
config_.track_id, config_.bitrate_priority});
}
void AudioSendStream::RemoveBitrateObserver() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
rtc::Event thread_sync_event;
worker_queue_->PostTask([this, &thread_sync_event] {
RTC_DCHECK_RUN_ON(worker_queue_);
registered_with_allocator_ = false;
bitrate_allocator_->RemoveObserver(this);
thread_sync_event.Set();
});

10
audio/audio_send_stream.h
View File

@ -122,9 +122,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
static void ReconfigureBitrateObserver(AudioSendStream* stream,
const Config& new_config);
void ConfigureBitrateObserver(int min_bitrate_bps,
int max_bitrate_bps,
double bitrate_priority);
void ConfigureBitrateObserver() RTC_RUN_ON(worker_queue_);
void RemoveBitrateObserver();
// Sets per-packet overhead on encoded (for ANA) based on current known values
@ -153,7 +151,8 @@ class AudioSendStream final : public webrtc::AudioSendStream,
size_t encoder_num_channels_ = 0;
bool sending_ = false;
BitrateAllocatorInterface* const bitrate_allocator_;
BitrateAllocatorInterface* const bitrate_allocator_
RTC_GUARDED_BY(worker_queue_);
RtpTransportControllerSendInterface* const rtp_transport_;
rtc::CriticalSection packet_loss_tracker_cs_;
@ -187,6 +186,9 @@ class AudioSendStream final : public webrtc::AudioSendStream,
size_t audio_overhead_per_packet_bytes_
RTC_GUARDED_BY(overhead_per_packet_lock_) = 0;
bool registered_with_allocator_ RTC_GUARDED_BY(worker_queue_) = false;
size_t total_packet_overhead_bytes_ RTC_GUARDED_BY(worker_queue_) = 0;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
};
} // namespace internal