Remove SetLatency/GetLatency from MediaSourceInterface API level
Bug: webrtc:10287
Change-Id: I74fad31db98b75791085688438064f9510b0b6fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133165
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27692}diff --git a/api/media_stream_interface.cc b/api/media_stream_interface.cc
index cf994cb..73566c4 100644
--- a/api/media_stream_interface.cc
+++ b/api/media_stream_interface.cc
@@ -32,8 +32,4 @@
return {};
}
-double MediaSourceInterface::GetLatency() const {
- return 0.0;
-}
-
} // namespace webrtc
diff --git a/api/media_stream_interface.h b/api/media_stream_interface.h
index ccacb4c..f4ea4a6 100644
--- a/api/media_stream_interface.h
+++ b/api/media_stream_interface.h
@@ -62,13 +62,6 @@
virtual bool remote() const = 0;
- // Sets the minimum latency of the remote source until audio playout. Actual
- // observered latency may differ depending on the source. |latency| is in the
- // range of [0.0, 10.0] seconds.
- // TODO(kuddai) make pure virtual once not only remote tracks support latency.
- virtual void SetLatency(double latency) {}
- virtual double GetLatency() const;
-
protected:
~MediaSourceInterface() override = default;
};
diff --git a/api/rtp_receiver_interface.cc b/api/rtp_receiver_interface.cc
index 615a31d..52f72df 100644
--- a/api/rtp_receiver_interface.cc
+++ b/api/rtp_receiver_interface.cc
@@ -58,7 +58,4 @@
return nullptr;
}
-void RtpReceiverInterface::SetJitterBufferMinimumDelay(
- absl::optional<double> delay_seconds) {}
-
} // namespace webrtc
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index d1ef137..3934215 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -131,10 +131,9 @@
// Sets the jitter buffer minimum delay until media playout. Actual observed
// delay may differ depending on the congestion control. |delay_seconds| is a
// positive value including 0.0 measured in seconds. |nullopt| means default
- // value must be used. TODO(kuddai): remove the default implmenetation once
- // the subclasses in Chromium implement this.
+ // value must be used.
virtual void SetJitterBufferMinimumDelay(
- absl::optional<double> delay_seconds);
+ absl::optional<double> delay_seconds) = 0;
// TODO(zhihuang): Remove the default implementation once the subclasses
// implement this. Currently, the only relevant subclass is the
diff --git a/api/video_track_source_proxy.h b/api/video_track_source_proxy.h
index eb11bef..820cdcb 100644
--- a/api/video_track_source_proxy.h
+++ b/api/video_track_source_proxy.h
@@ -32,8 +32,6 @@
rtc::VideoSinkInterface<VideoFrame>*,
const rtc::VideoSinkWants&)
PROXY_WORKER_METHOD1(void, RemoveSink, rtc::VideoSinkInterface<VideoFrame>*)
-PROXY_WORKER_METHOD1(void, SetLatency, double)
-PROXY_WORKER_CONSTMETHOD0(double, GetLatency)
PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
END_PROXY_MAP()
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 5ff4b16..16973e4 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -135,6 +135,10 @@
"dtmf_sender.h",
"ice_server_parsing.cc",
"ice_server_parsing.h",
+ "jitter_buffer_delay.cc",
+ "jitter_buffer_delay.h",
+ "jitter_buffer_delay_interface.h",
+ "jitter_buffer_delay_proxy.h",
"jsep_ice_candidate.cc",
"jsep_session_description.cc",
"local_audio_source.cc",
@@ -149,10 +153,6 @@
"peer_connection_factory.cc",
"peer_connection_factory.h",
"peer_connection_internal.h",
- "playout_latency.cc",
- "playout_latency.h",
- "playout_latency_interface.h",
- "playout_latency_proxy.h",
"remote_audio_source.cc",
"remote_audio_source.h",
"rtc_stats_collector.cc",
@@ -447,6 +447,7 @@
"data_channel_unittest.cc",
"dtmf_sender_unittest.cc",
"ice_server_parsing_unittest.cc",
+ "jitter_buffer_delay_unittest.cc",
"jsep_session_description_unittest.cc",
"local_audio_source_unittest.cc",
"media_stream_unittest.cc",
@@ -466,7 +467,6 @@
"peer_connection_simulcast_unittest.cc",
"peer_connection_wrapper.cc",
"peer_connection_wrapper.h",
- "playout_latency_unittest.cc",
"proxy_unittest.cc",
"rtc_stats_collector_unittest.cc",
"rtc_stats_integrationtest.cc",
diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc
index 61ea340..a470707 100644
--- a/pc/audio_rtp_receiver.cc
+++ b/pc/audio_rtp_receiver.cc
@@ -17,6 +17,8 @@
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "pc/audio_track.h"
+#include "pc/jitter_buffer_delay.h"
+#include "pc/jitter_buffer_delay_proxy.h"
#include "pc/media_stream.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
@@ -25,10 +27,6 @@
namespace webrtc {
-namespace {
-constexpr double kDefaultLatency = 0.0;
-} // namespace
-
AudioRtpReceiver::AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids)
@@ -46,7 +44,11 @@
track_(AudioTrackProxy::Create(rtc::Thread::Current(),
AudioTrack::Create(receiver_id, source_))),
cached_track_enabled_(track_->enabled()),
- attachment_id_(GenerateUniqueId()) {
+ attachment_id_(GenerateUniqueId()),
+ delay_(JitterBufferDelayProxy::Create(
+ rtc::Thread::Current(),
+ worker_thread_,
+ new rtc::RefCountedObject<JitterBufferDelay>(worker_thread))) {
RTC_DCHECK(worker_thread_);
RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
@@ -162,9 +164,11 @@
}
if (ssrc_) {
source_->Stop(media_channel_, *ssrc_);
+ delay_->OnStop();
}
ssrc_ = ssrc;
source_->Start(media_channel_, *ssrc_);
+ delay_->OnStart(media_channel_, *ssrc_);
Reconfigure();
}
@@ -238,7 +242,7 @@
void AudioRtpReceiver::SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds) {
- source_->SetLatency(delay_seconds.value_or(kDefaultLatency));
+ delay_->Set(delay_seconds);
}
void AudioRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h
index 9ad5c61..bc80fbe 100644
--- a/pc/audio_rtp_receiver.h
+++ b/pc/audio_rtp_receiver.h
@@ -22,6 +22,7 @@
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "media/base/media_channel.h"
+#include "pc/jitter_buffer_delay_interface.h"
#include "pc/remote_audio_source.h"
#include "pc/rtp_receiver.h"
#include "rtc_base/ref_counted_object.h"
@@ -122,6 +123,9 @@
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
+ // Allows to thread safely change playout delay. Handles caching cases if
+ // |SetJitterBufferMinimumDelay| is called before start.
+ rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
};
} // namespace webrtc
diff --git a/pc/jitter_buffer_delay.cc b/pc/jitter_buffer_delay.cc
new file mode 100644
index 0000000..c9506b3
--- /dev/null
+++ b/pc/jitter_buffer_delay.cc
@@ -0,0 +1,67 @@
+/*
+ * Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "pc/jitter_buffer_delay.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/location.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_checker.h"
+
+namespace {
+constexpr int kDefaultDelay = 0;
+constexpr int kMaximumDelayMs = 10000;
+} // namespace
+
+namespace webrtc {
+
+JitterBufferDelay::JitterBufferDelay(rtc::Thread* worker_thread)
+ : signaling_thread_(rtc::Thread::Current()), worker_thread_(worker_thread) {
+ RTC_DCHECK(worker_thread_);
+}
+
+void JitterBufferDelay::OnStart(cricket::Delayable* media_channel,
+ uint32_t ssrc) {
+ RTC_DCHECK_RUN_ON(signaling_thread_);
+
+ media_channel_ = media_channel;
+ ssrc_ = ssrc;
+
+ // Trying to apply cached delay for the audio stream.
+ if (cached_delay_seconds_) {
+ Set(cached_delay_seconds_.value());
+ }
+}
+
+void JitterBufferDelay::OnStop() {
+ RTC_DCHECK_RUN_ON(signaling_thread_);
+ // Assume that audio stream is no longer present.
+ media_channel_ = nullptr;
+ ssrc_ = absl::nullopt;
+}
+
+void JitterBufferDelay::Set(absl::optional<double> delay_seconds) {
+ RTC_DCHECK_RUN_ON(worker_thread_);
+
+ // TODO(kuddai) propagate absl::optional deeper down as default preference.
+ int delay_ms =
+ rtc::saturated_cast<int>(delay_seconds.value_or(kDefaultDelay) * 1000);
+ delay_ms = rtc::SafeClamp(delay_ms, 0, kMaximumDelayMs);
+
+ cached_delay_seconds_ = delay_seconds;
+ if (media_channel_ && ssrc_) {
+ media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms);
+ }
+}
+
+} // namespace webrtc
diff --git a/pc/jitter_buffer_delay.h b/pc/jitter_buffer_delay.h
new file mode 100644
index 0000000..8edfc6c
--- /dev/null
+++ b/pc/jitter_buffer_delay.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef PC_JITTER_BUFFER_DELAY_H_
+#define PC_JITTER_BUFFER_DELAY_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "media/base/delayable.h"
+#include "pc/jitter_buffer_delay_interface.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+// JitterBufferDelay converts delay from seconds to milliseconds for the
+// underlying media channel. It also handles cases when user sets delay before
+// the start of media_channel by caching its request. Note, this class is not
+// thread safe. Its thread safe version is defined in
+// pc/jitter_buffer_delay_proxy.h
+class JitterBufferDelay : public JitterBufferDelayInterface {
+ public:
+ // Must be called on signaling thread.
+ explicit JitterBufferDelay(rtc::Thread* worker_thread);
+
+ void OnStart(cricket::Delayable* media_channel, uint32_t ssrc) override;
+
+ void OnStop() override;
+
+ void Set(absl::optional<double> delay_seconds) override;
+
+ private:
+ // Throughout webrtc source, sometimes it is also called as |main_thread_|.
+ rtc::Thread* const signaling_thread_;
+ rtc::Thread* const worker_thread_;
+ // Media channel and ssrc together uniqely identify audio stream.
+ cricket::Delayable* media_channel_ = nullptr;
+ absl::optional<uint32_t> ssrc_;
+ absl::optional<double> cached_delay_seconds_;
+};
+
+} // namespace webrtc
+
+#endif // PC_JITTER_BUFFER_DELAY_H_
diff --git a/pc/jitter_buffer_delay_interface.h b/pc/jitter_buffer_delay_interface.h
new file mode 100644
index 0000000..f2132d3
--- /dev/null
+++ b/pc/jitter_buffer_delay_interface.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef PC_JITTER_BUFFER_DELAY_INTERFACE_H_
+#define PC_JITTER_BUFFER_DELAY_INTERFACE_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "media/base/delayable.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// JitterBufferDelay delivers user's queries to the underlying media channel. It
+// can describe either video or audio delay for receiving stream. "Interface"
+// suffix in the interface name is required to be compatible with api/proxy.cc
+class JitterBufferDelayInterface : public rtc::RefCountInterface {
+ public:
+ // OnStart allows to uniqely identify to which receiving stream playout
+ // delay must correpond through |media_channel| and |ssrc| pair.
+ virtual void OnStart(cricket::Delayable* media_channel, uint32_t ssrc) = 0;
+
+ // Indicates that underlying receiving stream is stopped.
+ virtual void OnStop() = 0;
+
+ virtual void Set(absl::optional<double> delay_seconds) = 0;
+};
+
+} // namespace webrtc
+
+#endif // PC_JITTER_BUFFER_DELAY_INTERFACE_H_
diff --git a/pc/playout_latency_proxy.h b/pc/jitter_buffer_delay_proxy.h
similarity index 70%
rename from pc/playout_latency_proxy.h
rename to pc/jitter_buffer_delay_proxy.h
index 22f02c5..b3380fd 100644
--- a/pc/playout_latency_proxy.h
+++ b/pc/jitter_buffer_delay_proxy.h
@@ -8,25 +8,24 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef PC_PLAYOUT_LATENCY_PROXY_H_
-#define PC_PLAYOUT_LATENCY_PROXY_H_
+#ifndef PC_JITTER_BUFFER_DELAY_PROXY_H_
+#define PC_JITTER_BUFFER_DELAY_PROXY_H_
#include <stdint.h>
#include "api/proxy.h"
#include "media/base/delayable.h"
-#include "pc/playout_latency_interface.h"
+#include "pc/jitter_buffer_delay_interface.h"
namespace webrtc {
-BEGIN_PROXY_MAP(PlayoutLatency)
+BEGIN_PROXY_MAP(JitterBufferDelay)
PROXY_SIGNALING_THREAD_DESTRUCTOR()
PROXY_METHOD2(void, OnStart, cricket::Delayable*, uint32_t)
PROXY_METHOD0(void, OnStop)
-PROXY_WORKER_METHOD1(void, SetLatency, double)
-PROXY_WORKER_CONSTMETHOD0(double, GetLatency)
+PROXY_WORKER_METHOD1(void, Set, absl::optional<double>)
END_PROXY_MAP()
} // namespace webrtc
-#endif // PC_PLAYOUT_LATENCY_PROXY_H_
+#endif // PC_JITTER_BUFFER_DELAY_PROXY_H_
diff --git a/pc/jitter_buffer_delay_unittest.cc b/pc/jitter_buffer_delay_unittest.cc
new file mode 100644
index 0000000..383a4b7
--- /dev/null
+++ b/pc/jitter_buffer_delay_unittest.cc
@@ -0,0 +1,90 @@
+/*
+ * Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "pc/jitter_buffer_delay.h"
+#include "pc/test/mock_delayable.h"
+#include "rtc_base/ref_counted_object.h"
+#include "rtc_base/thread.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::Return;
+
+namespace {
+constexpr int kSsrc = 1234;
+} // namespace
+
+namespace webrtc {
+
+class JitterBufferDelayTest : public ::testing::Test {
+ public:
+ JitterBufferDelayTest()
+ : delay_(new rtc::RefCountedObject<JitterBufferDelay>(
+ rtc::Thread::Current())) {}
+
+ protected:
+ rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
+ MockDelayable delayable_;
+};
+
+TEST_F(JitterBufferDelayTest, Set) {
+ delay_->OnStart(&delayable_, kSsrc);
+
+ EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 3000))
+ .WillOnce(Return(true));
+
+ // Delay in seconds.
+ delay_->Set(3.0);
+}
+
+TEST_F(JitterBufferDelayTest, Caching) {
+ // Check that value is cached before start.
+ delay_->Set(4.0);
+
+ // Check that cached value applied on the start.
+ EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 4000))
+ .WillOnce(Return(true));
+ delay_->OnStart(&delayable_, kSsrc);
+}
+
+TEST_F(JitterBufferDelayTest, Clamping) {
+ delay_->OnStart(&delayable_, kSsrc);
+
+ // In current Jitter Buffer implementation (Audio or Video) maximum supported
+ // value is 10000 milliseconds.
+ EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 10000))
+ .WillOnce(Return(true));
+ delay_->Set(10.5);
+
+ // Test int overflow.
+ EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 10000))
+ .WillOnce(Return(true));
+ delay_->Set(21474836470.0);
+
+ EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
+ .WillOnce(Return(true));
+ delay_->Set(-21474836470.0);
+
+ // Boundary value in seconds to milliseconds conversion.
+ EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
+ .WillOnce(Return(true));
+ delay_->Set(0.0009);
+
+ EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
+ .WillOnce(Return(true));
+
+ delay_->Set(-2.0);
+}
+
+} // namespace webrtc
diff --git a/pc/playout_latency.cc b/pc/playout_latency.cc
deleted file mode 100644
index 063f232..0000000
--- a/pc/playout_latency.cc
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright 2019 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "pc/playout_latency.h"
-
-#include "rtc_base/checks.h"
-#include "rtc_base/location.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/numerics/safe_conversions.h"
-#include "rtc_base/numerics/safe_minmax.h"
-#include "rtc_base/thread.h"
-#include "rtc_base/thread_checker.h"
-
-namespace {
-constexpr int kDefaultLatency = 0;
-constexpr int kMaximumDelayMs = 10000;
-} // namespace
-
-namespace webrtc {
-
-PlayoutLatency::PlayoutLatency(rtc::Thread* worker_thread)
- : signaling_thread_(rtc::Thread::Current()), worker_thread_(worker_thread) {
- RTC_DCHECK(worker_thread_);
-}
-
-void PlayoutLatency::OnStart(cricket::Delayable* media_channel, uint32_t ssrc) {
- RTC_DCHECK_RUN_ON(signaling_thread_);
-
- media_channel_ = media_channel;
- ssrc_ = ssrc;
-
- // Trying to apply cached latency for the audio stream.
- if (cached_latency_) {
- SetLatency(cached_latency_.value());
- }
-}
-
-void PlayoutLatency::OnStop() {
- RTC_DCHECK_RUN_ON(signaling_thread_);
- // Assume that audio stream is no longer present for latency calls.
- media_channel_ = nullptr;
- ssrc_ = absl::nullopt;
-}
-
-void PlayoutLatency::SetLatency(double latency) {
- RTC_DCHECK_RUN_ON(worker_thread_);
-
- int delay_ms = rtc::dchecked_cast<int>(latency * 1000);
- delay_ms = rtc::SafeClamp(delay_ms, 0, kMaximumDelayMs);
-
- cached_latency_ = latency;
- if (media_channel_ && ssrc_) {
- media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms);
- }
-}
-
-double PlayoutLatency::GetLatency() const {
- RTC_DCHECK_RUN_ON(worker_thread_);
-
- absl::optional<int> delay_ms;
- if (media_channel_ && ssrc_) {
- delay_ms = media_channel_->GetBaseMinimumPlayoutDelayMs(ssrc_.value());
- }
-
- if (delay_ms) {
- return delay_ms.value() / 1000.0;
- } else {
- return cached_latency_.value_or(kDefaultLatency);
- }
-}
-
-} // namespace webrtc
diff --git a/pc/playout_latency.h b/pc/playout_latency.h
deleted file mode 100644
index f51a927..0000000
--- a/pc/playout_latency.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright 2019 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef PC_PLAYOUT_LATENCY_H_
-#define PC_PLAYOUT_LATENCY_H_
-
-#include <stdint.h>
-
-#include "absl/types/optional.h"
-#include "media/base/delayable.h"
-#include "pc/playout_latency_interface.h"
-#include "rtc_base/thread.h"
-
-namespace webrtc {
-
-// PlayoutLatency converts latency measured in seconds to delay measured in
-// milliseconds for the underlying media channel. It also handles cases when
-// user sets Latency before the start of media_channel by caching its request.
-// Note, this class is not thread safe. Its thread safe version is defined in
-// pc/playout_latency_proxy.h
-class PlayoutLatency : public PlayoutLatencyInterface {
- public:
- // Must be called on signaling thread.
- explicit PlayoutLatency(rtc::Thread* worker_thread);
-
- void OnStart(cricket::Delayable* media_channel, uint32_t ssrc) override;
-
- void OnStop() override;
-
- void SetLatency(double latency) override;
-
- double GetLatency() const override;
-
- private:
- // Throughout webrtc source, sometimes it is also called as |main_thread_|.
- rtc::Thread* const signaling_thread_;
- rtc::Thread* const worker_thread_;
- // Media channel and ssrc together uniqely identify audio stream.
- cricket::Delayable* media_channel_ = nullptr;
- absl::optional<uint32_t> ssrc_;
- absl::optional<double> cached_latency_;
-};
-
-} // namespace webrtc
-
-#endif // PC_PLAYOUT_LATENCY_H_
diff --git a/pc/playout_latency_interface.h b/pc/playout_latency_interface.h
deleted file mode 100644
index e995733..0000000
--- a/pc/playout_latency_interface.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright 2019 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef PC_PLAYOUT_LATENCY_INTERFACE_H_
-#define PC_PLAYOUT_LATENCY_INTERFACE_H_
-
-#include <stdint.h>
-
-#include "media/base/delayable.h"
-#include "rtc_base/ref_count.h"
-
-namespace webrtc {
-
-// PlayoutLatency delivers user's latency queries to the underlying media
-// channel. It can describe either video or audio latency for receiving stream.
-// "Interface" suffix in the interface name is required to be compatible with
-// api/proxy.cc
-class PlayoutLatencyInterface : public rtc::RefCountInterface {
- public:
- // OnStart allows to uniqely identify to which receiving stream playout
- // latency must correpond through |media_channel| and |ssrc| pair.
- virtual void OnStart(cricket::Delayable* media_channel, uint32_t ssrc) = 0;
-
- // Indicates that underlying receiving stream is stopped.
- virtual void OnStop() = 0;
-
- // Sets latency in seconds.
- virtual void SetLatency(double latency) = 0;
-
- // Returns latency in seconds.
- virtual double GetLatency() const = 0;
-};
-
-} // namespace webrtc
-
-#endif // PC_PLAYOUT_LATENCY_INTERFACE_H_
diff --git a/pc/playout_latency_unittest.cc b/pc/playout_latency_unittest.cc
deleted file mode 100644
index db139f4..0000000
--- a/pc/playout_latency_unittest.cc
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- * Copyright 2019 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdint.h>
-
-#include "absl/types/optional.h"
-#include "api/scoped_refptr.h"
-#include "pc/playout_latency.h"
-#include "pc/test/mock_delayable.h"
-#include "rtc_base/ref_counted_object.h"
-#include "rtc_base/thread.h"
-#include "test/gmock.h"
-#include "test/gtest.h"
-
-using ::testing::Return;
-
-namespace {
-constexpr int kSsrc = 1234;
-} // namespace
-
-namespace webrtc {
-
-class PlayoutLatencyTest : public ::testing::Test {
- public:
- PlayoutLatencyTest()
- : latency_(
- new rtc::RefCountedObject<PlayoutLatency>(rtc::Thread::Current())) {
- }
-
- protected:
- rtc::scoped_refptr<PlayoutLatencyInterface> latency_;
- MockDelayable delayable_;
-};
-
-TEST_F(PlayoutLatencyTest, DefaultValue) {
- EXPECT_DOUBLE_EQ(0.0, latency_->GetLatency());
-}
-
-TEST_F(PlayoutLatencyTest, GetLatency) {
- latency_->OnStart(&delayable_, kSsrc);
-
- EXPECT_CALL(delayable_, GetBaseMinimumPlayoutDelayMs(kSsrc))
- .WillOnce(Return(2000));
- // Latency in seconds.
- EXPECT_DOUBLE_EQ(2.0, latency_->GetLatency());
-
- EXPECT_CALL(delayable_, GetBaseMinimumPlayoutDelayMs(kSsrc))
- .WillOnce(Return(absl::nullopt));
- // When no value is returned by GetBaseMinimumPlayoutDelayMs, and there are
- // no caching, then return default value.
- EXPECT_DOUBLE_EQ(0.0, latency_->GetLatency());
-}
-
-TEST_F(PlayoutLatencyTest, SetLatency) {
- latency_->OnStart(&delayable_, kSsrc);
-
- EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 3000))
- .WillOnce(Return(true));
-
- // Latency in seconds.
- latency_->SetLatency(3.0);
-}
-
-TEST_F(PlayoutLatencyTest, Caching) {
- // Check that value is cached before start.
- latency_->SetLatency(4.0);
- // Latency in seconds.
- EXPECT_DOUBLE_EQ(4.0, latency_->GetLatency());
-
- // Check that cached value applied on the start.
- EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 4000))
- .WillOnce(Return(true));
- latency_->OnStart(&delayable_, kSsrc);
-
- EXPECT_CALL(delayable_, GetBaseMinimumPlayoutDelayMs(kSsrc))
- .WillOnce(Return(absl::nullopt));
- // On false the latest cached value is returned.
- EXPECT_DOUBLE_EQ(4.0, latency_->GetLatency());
-
- latency_->OnStop();
-
- // Check that after stop it returns last cached value.
- EXPECT_DOUBLE_EQ(4.0, latency_->GetLatency());
-}
-
-TEST_F(PlayoutLatencyTest, Clamping) {
- latency_->OnStart(&delayable_, kSsrc);
-
- // In current Jitter Buffer implementation (Audio or Video) maximum supported
- // value is 10000 milliseconds.
- EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 10000))
- .WillOnce(Return(true));
- latency_->SetLatency(10.5);
-
- // Boundary value in seconds to milliseconds conversion.
- EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
- .WillOnce(Return(true));
- latency_->SetLatency(0.0009);
-
- EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
- .WillOnce(Return(true));
-
- latency_->SetLatency(-2.0);
-}
-
-} // namespace webrtc
diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc
index aba4400..ed8ef2d 100644
--- a/pc/remote_audio_source.cc
+++ b/pc/remote_audio_source.cc
@@ -16,8 +16,6 @@
#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"
@@ -52,11 +50,7 @@
RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread)
: main_thread_(rtc::Thread::Current()),
worker_thread_(worker_thread),
- state_(MediaSourceInterface::kLive),
- latency_(PlayoutLatencyProxy::Create(
- main_thread_,
- worker_thread_,
- new rtc::RefCountedObject<PlayoutLatency>(worker_thread))) {
+ state_(MediaSourceInterface::kLive) {
RTC_DCHECK(main_thread_);
RTC_DCHECK(worker_thread_);
}
@@ -79,9 +73,6 @@
media_channel->SetRawAudioSink(ssrc,
absl::make_unique<AudioDataProxy>(this));
});
-
- // Apply latency to the audio stream if |SetLatency| was called before.
- latency_->OnStart(media_channel, ssrc);
}
void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel,
@@ -89,8 +80,6 @@
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
- latency_->OnStop();
-
worker_thread_->Invoke<void>(
RTC_FROM_HERE, [&] { media_channel->SetRawAudioSink(ssrc, nullptr); });
}
@@ -113,14 +102,6 @@
}
}
-void RemoteAudioSource::SetLatency(double latency) {
- latency_->SetLatency(latency);
-}
-
-double RemoteAudioSource::GetLatency() const {
- return latency_->GetLatency();
-}
-
void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer));
diff --git a/pc/remote_audio_source.h b/pc/remote_audio_source.h
index f4a5ec8..399e7e3 100644
--- a/pc/remote_audio_source.h
+++ b/pc/remote_audio_source.h
@@ -17,7 +17,6 @@
#include "api/call/audio_sink.h"
#include "api/notifier.h"
#include "pc/channel.h"
-#include "pc/playout_latency_interface.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/message_handler.h"
@@ -47,8 +46,6 @@
// AudioSourceInterface implementation.
void SetVolume(double volume) override;
- void SetLatency(double latency) override;
- double GetLatency() const override;
void RegisterAudioObserver(AudioObserver* observer) override;
void UnregisterAudioObserver(AudioObserver* observer) override;
@@ -72,9 +69,6 @@
rtc::CriticalSection sink_lock_;
std::list<AudioTrackSinkInterface*> sinks_;
SourceState state_;
- // Allows to thread safely change playout latency. Handles caching cases if
- // |SetLatency| is called before start.
- rtc::scoped_refptr<PlayoutLatencyInterface> latency_;
};
} // namespace webrtc
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index 67cbe2c..90ca03f 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -459,41 +459,6 @@
RunSetLastLayerAsInactiveTest(video_rtp_sender_.get());
}
- void VerifyTrackLatencyBehaviour(cricket::Delayable* media_channel,
- MediaStreamTrackInterface* track,
- MediaSourceInterface* source,
- uint32_t ssrc) {
- absl::optional<int> delay_ms; // In milliseconds.
- double latency_s = 0.5; // In seconds.
-
- source->SetLatency(latency_s);
- delay_ms = media_channel->GetBaseMinimumPlayoutDelayMs(ssrc);
- EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
-
- // Disabling the track should take no effect on previously set value.
- track->set_enabled(false);
- delay_ms = media_channel->GetBaseMinimumPlayoutDelayMs(ssrc);
- 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;
- source->SetLatency(latency_s);
- delay_ms = media_channel->GetBaseMinimumPlayoutDelayMs(ssrc);
- EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
-
- // Enabling the track should take no effect on previously set value.
- track->set_enabled(true);
- delay_ms = media_channel->GetBaseMinimumPlayoutDelayMs(ssrc);
- EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
-
- // We still should be able to change latency.
- latency_s = 0.0;
- source->SetLatency(latency_s);
- delay_ms = media_channel->GetBaseMinimumPlayoutDelayMs(ssrc);
- EXPECT_EQ(0, delay_ms.value_or(-1));
- EXPECT_DOUBLE_EQ(latency_s, delay_ms.value_or(0) / 1000.0);
- }
-
// Check that minimum Jitter Buffer delay is propagated to the underlying
// |media_channel|.
void VerifyRtpReceiverDelayBehaviour(cricket::Delayable* media_channel,
@@ -687,44 +652,10 @@
DestroyAudioRtpReceiver();
}
-TEST_F(RtpSenderReceiverTest, RemoteAudioSourceLatency) {
- absl::optional<int> delay_ms; // 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, 300);
-
- // Check that calling GetLatency on the source that hasn't been started yet
- // won't trigger caching and return default value.
- EXPECT_DOUBLE_EQ(source->GetLatency(), 0);
-
- // Check that cached latency will be applied on start.
- source->SetLatency(0.4);
- EXPECT_DOUBLE_EQ(source->GetLatency(), 0.4);
- source->Start(voice_media_channel_, kAudioSsrc);
- delay_ms = voice_media_channel_->GetBaseMinimumPlayoutDelayMs(kAudioSsrc);
- EXPECT_EQ(400, delay_ms);
-}
-
-TEST_F(RtpSenderReceiverTest, RemoteAudioTrackLatency) {
- CreateAudioRtpReceiver();
- VerifyTrackLatencyBehaviour(voice_media_channel_, audio_track_.get(),
- audio_track_->GetSource(), kAudioSsrc);
-}
-
-TEST_F(RtpSenderReceiverTest, RemoteVideoTrackLatency) {
- CreateVideoRtpReceiver();
- VerifyTrackLatencyBehaviour(video_media_channel_, video_track_.get(),
- video_track_->GetSource(), kVideoSsrc);
-}
-
TEST_F(RtpSenderReceiverTest, AudioRtpReceiverDelay) {
CreateAudioRtpReceiver();
VerifyRtpReceiverDelayBehaviour(voice_media_channel_,
audio_rtp_receiver_.get(), kAudioSsrc);
- VerifyTrackLatencyBehaviour(voice_media_channel_, audio_track_.get(),
- audio_track_->GetSource(), kAudioSsrc);
}
TEST_F(RtpSenderReceiverTest, VideoRtpReceiverDelay) {
diff --git a/pc/video_rtp_receiver.cc b/pc/video_rtp_receiver.cc
index 5a2e778..7a9ec98 100644
--- a/pc/video_rtp_receiver.cc
+++ b/pc/video_rtp_receiver.cc
@@ -17,6 +17,8 @@
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "api/video_track_source_proxy.h"
+#include "pc/jitter_buffer_delay.h"
+#include "pc/jitter_buffer_delay_proxy.h"
#include "pc/media_stream.h"
#include "pc/video_track.h"
#include "rtc_base/checks.h"
@@ -26,10 +28,6 @@
namespace webrtc {
-namespace {
-constexpr double kDefaultLatency = 0.0;
-} // namespace
-
VideoRtpReceiver::VideoRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids)
@@ -43,7 +41,7 @@
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams)
: worker_thread_(worker_thread),
id_(receiver_id),
- source_(new RefCountedObject<VideoRtpTrackSource>(worker_thread_)),
+ source_(new RefCountedObject<VideoRtpTrackSource>()),
track_(VideoTrackProxy::Create(
rtc::Thread::Current(),
worker_thread,
@@ -53,7 +51,11 @@
worker_thread,
source_),
worker_thread))),
- attachment_id_(GenerateUniqueId()) {
+ attachment_id_(GenerateUniqueId()),
+ delay_(JitterBufferDelayProxy::Create(
+ rtc::Thread::Current(),
+ worker_thread,
+ new rtc::RefCountedObject<JitterBufferDelay>(worker_thread))) {
RTC_DCHECK(worker_thread_);
SetStreams(streams);
source_->SetState(MediaSourceInterface::kLive);
@@ -127,7 +129,7 @@
// media channel has already been deleted.
SetSink(nullptr);
}
- source_->Stop();
+ delay_->OnStop();
stopped_ = true;
}
@@ -148,7 +150,7 @@
MaybeAttachFrameDecryptorToMediaChannel(
ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
- source_->Start(media_channel_, ssrc);
+ delay_->OnStart(media_channel_, ssrc);
}
void VideoRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
@@ -198,7 +200,7 @@
void VideoRtpReceiver::SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds) {
- source_->SetLatency(delay_seconds.value_or(kDefaultLatency));
+ delay_->Set(delay_seconds);
}
void VideoRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h
index 9ad4fe7..10354b0 100644
--- a/pc/video_rtp_receiver.h
+++ b/pc/video_rtp_receiver.h
@@ -27,8 +27,7 @@
#include "api/video/video_source_interface.h"
#include "media/base/media_channel.h"
#include "media/base/video_broadcaster.h"
-#include "pc/playout_latency.h"
-#include "pc/playout_latency_proxy.h"
+#include "pc/jitter_buffer_delay_interface.h"
#include "pc/rtp_receiver.h"
#include "pc/video_track_source.h"
#include "rtc_base/ref_counted_object.h"
@@ -108,42 +107,23 @@
std::vector<RtpSource> GetSources() const override;
+ private:
class VideoRtpTrackSource : public VideoTrackSource {
public:
- explicit VideoRtpTrackSource(rtc::Thread* worker_thread)
- : VideoTrackSource(true /* remote */),
- latency_(PlayoutLatencyProxy::Create(
- rtc::Thread::Current(),
- worker_thread,
- new rtc::RefCountedObject<PlayoutLatency>(worker_thread))) {}
+ VideoRtpTrackSource() : VideoTrackSource(true /* remote */) {}
rtc::VideoSourceInterface<VideoFrame>* source() override {
return &broadcaster_;
}
rtc::VideoSinkInterface<VideoFrame>* sink() { return &broadcaster_; }
- void SetLatency(double latency) override { latency_->SetLatency(latency); }
-
- void Start(cricket::VideoMediaChannel* media_channel, uint32_t ssrc) {
- latency_->OnStart(media_channel, ssrc);
- }
-
- void Stop() { latency_->OnStop(); }
-
- double GetLatency() const override { return latency_->GetLatency(); }
-
private:
- // Allows to thread safely change playout latency. Handles caching cases if
- // |SetLatency| is called before start.
- rtc::scoped_refptr<PlayoutLatencyInterface> latency_;
-
// |broadcaster_| is needed since the decoder can only handle one sink.
// It might be better if the decoder can handle multiple sinks and consider
// the VideoSinkWants.
rtc::VideoBroadcaster broadcaster_;
};
- private:
bool SetSink(rtc::VideoSinkInterface<VideoFrame>* sink);
rtc::Thread* const worker_thread_;
@@ -161,6 +141,9 @@
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
+ // Allows to thread safely change jitter buffer delay. Handles caching cases
+ // if |SetJitterBufferMinimumDelay| is called before start.
+ rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
};
} // namespace webrtc