Add latency to remote source api.
Latency corresponds to base minimum delay on NetEq.
Bug: webrtc:10287
Change-Id: I538d202e3e4fe07b779c46bf560e2fde38e0468e
Reviewed-on: https://webrtc-review.googlesource.com/c/121704
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26724}diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index b172460..1f015f6 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -48,6 +48,7 @@
#include "pc/dtls_srtp_transport.h"
#include "pc/local_audio_source.h"
#include "pc/media_stream.h"
+#include "pc/remote_audio_source.h"
#include "pc/rtp_receiver.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_transport_internal.h"
@@ -522,6 +523,103 @@
DestroyAudioRtpReceiver();
}
+TEST_F(RtpSenderReceiverTest, RemoteAudioSourceLatencyCaching) {
+ absl::optional<int> delay_ms; // In milliseconds.
+ double latency_s = 0.5; // In seconds.
+ rtc::scoped_refptr<RemoteAudioSource> source =
+ new rtc::RefCountedObject<RemoteAudioSource>(rtc::Thread::Current());
+
+ // Check default value.
+ EXPECT_DOUBLE_EQ(source->GetLatency(), 0.0);
+
+ // Check caching behaviour.
+ source->SetLatency(latency_s);
+ EXPECT_DOUBLE_EQ(source->GetLatency(), latency_s);
+
+ // Check that cached value applied on the start.
+ source->Start(voice_media_channel_, kAudioSsrc);
+ delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc);
+ EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
+
+ // Check that setting latency changes delay.
+ latency_s = 0.8;
+ source->SetLatency(latency_s);
+ delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc);
+ EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
+ EXPECT_DOUBLE_EQ(latency_s, source->GetLatency());
+
+ // Check that if underlying delay is changed then remote source will reflect
+ // it.
+ delay_ms = 300;
+ voice_media_channel_->SetBaseMinimumPlayoutDelayMs(kAudioSsrc,
+ delay_ms.value());
+ EXPECT_DOUBLE_EQ(source->GetLatency(), delay_ms.value() / 1000.0);
+
+ // Check that after stop we get last cached value.
+ source->Stop(voice_media_channel_, kAudioSsrc);
+ EXPECT_DOUBLE_EQ(latency_s, source->GetLatency());
+
+ // Check that if we start source again with new ssrc then cached value is
+ // applied.
+ source->Start(voice_media_channel_, kAudioSsrc2);
+ delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc2);
+ EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
+
+ // Check rounding behavior.
+ source->SetLatency(2 / 1000.0);
+ delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc2);
+ EXPECT_EQ(0, delay_ms.value_or(-1));
+ EXPECT_DOUBLE_EQ(0, source->GetLatency());
+}
+
+TEST_F(RtpSenderReceiverTest, RemoteAudioSourceLatencyNoCaching) {
+ int delay_ms = 300; // In milliseconds.
+ rtc::scoped_refptr<RemoteAudioSource> source =
+ new rtc::RefCountedObject<RemoteAudioSource>(rtc::Thread::Current());
+
+ // Set it to value different from default zero.
+ voice_media_channel_->SetBaseMinimumPlayoutDelayMs(kAudioSsrc, delay_ms);
+
+ // Check that calling GetLatency on the source that hasn't been started yet
+ // won't trigger caching.
+ EXPECT_DOUBLE_EQ(source->GetLatency(), 0);
+ source->Start(voice_media_channel_, kAudioSsrc);
+ EXPECT_DOUBLE_EQ(source->GetLatency(), delay_ms / 1000.0);
+}
+
+TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetLatency) {
+ CreateAudioRtpReceiver();
+
+ absl::optional<int> delay_ms; // In milliseconds.
+ double latency_s = 0.5; // In seconds.
+ audio_track_->GetSource()->SetLatency(latency_s);
+ delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc);
+ EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
+
+ // Disabling the track should take no effect on previously set value.
+ audio_track_->set_enabled(false);
+ delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc);
+ EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
+
+ // When the track is disabled, we still should be able to set latency.
+ latency_s = 0.3;
+ audio_track_->GetSource()->SetLatency(latency_s);
+ delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc);
+ EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
+
+ // Enabling the track should take no effect on previously set value.
+ audio_track_->set_enabled(true);
+ delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc);
+ EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
+
+ // We still should be able to change latency.
+ latency_s = 0.0;
+ audio_track_->GetSource()->SetLatency(latency_s);
+ delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc);
+ EXPECT_EQ(0, delay_ms.value_or(-1));
+ EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
+}
+
// Test that the media channel isn't enabled for sending if the audio sender
// doesn't have both a track and SSRC.
TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {