Propagate base minimum delay from video jitter buffer to webrtc/api.
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.
Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc
index 63944c6..aba4400 100644
--- a/pc/remote_audio_source.cc
+++ b/pc/remote_audio_source.cc
@@ -16,6 +16,8 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "api/scoped_refptr.h"
+#include "pc/playout_latency.h"
+#include "pc/playout_latency_proxy.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/location.h"
@@ -26,11 +28,6 @@
namespace webrtc {
-namespace {
-constexpr int kDefaultLatency = 0;
-constexpr int kRoundToZeroThresholdMs = 10;
-} // namespace
-
// This proxy is passed to the underlying media engine to receive audio data as
// they come in. The data will then be passed back up to the RemoteAudioSource
// which will fan it out to all the sinks that have been added to it.
@@ -55,7 +52,11 @@
RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread)
: main_thread_(rtc::Thread::Current()),
worker_thread_(worker_thread),
- state_(MediaSourceInterface::kLive) {
+ state_(MediaSourceInterface::kLive),
+ latency_(PlayoutLatencyProxy::Create(
+ main_thread_,
+ worker_thread_,
+ new rtc::RefCountedObject<PlayoutLatency>(worker_thread))) {
RTC_DCHECK(main_thread_);
RTC_DCHECK(worker_thread_);
}
@@ -70,12 +71,6 @@
uint32_t ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
- // Check that there are no consecutive start calls.
- RTC_DCHECK(!media_channel_ && !ssrc_);
-
- // Remember media channel ssrc pair for latency calls.
- media_channel_ = media_channel;
- ssrc_ = ssrc;
// Register for callbacks immediately before AddSink so that we always get
// notified when a channel goes out of scope (signaled when "AudioDataProxy"
@@ -85,10 +80,8 @@
absl::make_unique<AudioDataProxy>(this));
});
- // Trying to apply cached latency for the audio stream.
- if (cached_latency_) {
- SetLatency(cached_latency_.value());
- }
+ // Apply latency to the audio stream if |SetLatency| was called before.
+ latency_->OnStart(media_channel, ssrc);
}
void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel,
@@ -96,9 +89,7 @@
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
- // Assume that audio stream is no longer present for latency calls.
- media_channel_ = nullptr;
- ssrc_ = absl::nullopt;
+ latency_->OnStop();
worker_thread_->Invoke<void>(
RTC_FROM_HERE, [&] { media_channel->SetRawAudioSink(ssrc, nullptr); });
@@ -123,50 +114,11 @@
}
void RemoteAudioSource::SetLatency(double latency) {
- RTC_DCHECK_GE(latency, 0);
- RTC_DCHECK_LE(latency, 10);
-
- int delay_ms = rtc::dchecked_cast<int>(latency * 1000);
- // In NetEq 0 delay has special meaning of being unconstrained value that is
- // why we round delay to 0 if it is small enough during conversion from
- // latency.
- if (delay_ms <= kRoundToZeroThresholdMs) {
- delay_ms = 0;
- }
-
- cached_latency_ = latency;
- SetDelayMs(delay_ms);
+ latency_->SetLatency(latency);
}
double RemoteAudioSource::GetLatency() const {
- absl::optional<int> delay_ms = GetDelayMs();
-
- if (delay_ms) {
- return delay_ms.value() / 1000.0;
- } else {
- return cached_latency_.value_or(kDefaultLatency);
- }
-}
-
-bool RemoteAudioSource::SetDelayMs(int delay_ms) {
- if (!media_channel_ || !ssrc_) {
- return false;
- }
-
- worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
- media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms);
- });
- return true;
-}
-
-absl::optional<int> RemoteAudioSource::GetDelayMs() const {
- if (!media_channel_ || !ssrc_) {
- return absl::nullopt;
- }
-
- return worker_thread_->Invoke<absl::optional<int>>(RTC_FROM_HERE, [&] {
- return media_channel_->GetBaseMinimumPlayoutDelayMs(ssrc_.value());
- });
+ return latency_->GetLatency();
}
void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {