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VideoRtpReceiver & AudioRtpReceiver threading fixes.

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For implementations where the signaling and worker threads are not
the same thread, this significantly cuts down on Thread::Invoke()s that
would block the signaling thread while waiting for the worker thread.

For Audio and Video Rtp receivers, the following methods now do not
block the signaling thread:
* GetParameters
* SetJitterBufferMinimumDelay
* GetSources
* SetFrameDecryptor / GetFrameDecryptor
* SetDepacketizerToDecoderFrameTransformer

Importantly this change also makes the track() accessor accessible
directly from the application thread (bypassing the proxy) since
for receiver objects, the track object is const.

Other changes:

* Remove RefCountedObject inheritance, use make_ref_counted instead.
* Every member variable in the rtp receiver classes is now RTC_GUARDED
* Stop() now fully clears up worker thread state, and Stop() is
  consistently called before destruction. This means that there's one
  thread hop instead of at least 4 before (sometimes more), per receiver.
* OnChanged triggered volume for audio tracks is done asynchronously.
* Deleted most of the JitterBufferDelay implementation. Turns out that
  it was largely unnecessary overhead and complexity.

It seems that these two classes are copy/pasted to a large extent
so further refactoring would be good in the future, as to not have to
fix each issue twice.

Bug: chromium:1184611
Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34022}
This commit is contained in:
Tommi 2021-05-17 14:50:10 +02:00 committed by WebRTC LUCI CQ
parent b27a9f9481
commit 4ccdf932e1
21 changed files with 585 additions and 583 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

22
api/rtp_receiver_interface.h
View file

@ -100,11 +100,13 @@ class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
// before it is sent across the network. This will decrypt the entire frame
// using the user provided decryption mechanism regardless of whether SRTP is
// enabled or not.
// TODO(bugs.webrtc.org/12772): Remove.
virtual void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
// Returns a pointer to the frame decryptor set previously by the
// user. This can be used to update the state of the object.
// TODO(bugs.webrtc.org/12772): Remove.
virtual rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const;
// Sets a frame transformer between the depacketizer and the decoder to enable
@ -120,25 +122,29 @@ class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
// Define proxy for RtpReceiverInterface.
// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
// are called on is an implementation detail.
BEGIN_PRIMARY_PROXY_MAP(RtpReceiver)
BEGIN_PROXY_MAP(RtpReceiver)
PROXY_PRIMARY_THREAD_DESTRUCTOR()
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
BYPASS_PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
streams)
BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
BYPASS_PROXY_CONSTMETHOD0(std::string, id)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
PROXY_SECONDARY_CONSTMETHOD0(RtpParameters, GetParameters)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
PROXY_METHOD1(void, SetJitterBufferMinimumDelay, absl::optional<double>)
PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources)
PROXY_METHOD1(void,
PROXY_SECONDARY_METHOD1(void,
SetJitterBufferMinimumDelay,
absl::optional<double>)
PROXY_SECONDARY_CONSTMETHOD0(std::vector<RtpSource>, GetSources)
// TODO(bugs.webrtc.org/12772): Remove.
PROXY_SECONDARY_METHOD1(void,
SetFrameDecryptor,
rtc::scoped_refptr<FrameDecryptorInterface>)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameDecryptorInterface>,
// TODO(bugs.webrtc.org/12772): Remove.
PROXY_SECONDARY_CONSTMETHOD0(rtc::scoped_refptr<FrameDecryptorInterface>,
GetFrameDecryptor)
PROXY_METHOD1(void,
PROXY_SECONDARY_METHOD1(void,
SetDepacketizerToDecoderFrameTransformer,
rtc::scoped_refptr<FrameTransformerInterface>)
END_PROXY_MAP()

50
pc/BUILD.gn
View file

@ -216,8 +216,6 @@ rtc_library("peerconnection") {
":connection_context",
":dtmf_sender",
":jitter_buffer_delay",
":jitter_buffer_delay_interface",
":jitter_buffer_delay_proxy",
":media_protocol_names",
":media_stream",
":peer_connection_message_handler",
@ -502,8 +500,6 @@ rtc_library("audio_rtp_receiver") {
deps = [
":audio_track",
":jitter_buffer_delay",
":jitter_buffer_delay_interface",
":jitter_buffer_delay_proxy",
":media_stream",
":remote_audio_source",
":rtp_receiver",
@ -520,6 +516,9 @@ rtc_library("audio_rtp_receiver") {
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:threading",
"../rtc_base/system:no_unique_address",
"../rtc_base/task_utils:pending_task_safety_flag",
"../rtc_base/task_utils:to_queued_task",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
@ -535,8 +534,6 @@ rtc_library("video_rtp_receiver") {
]
deps = [
":jitter_buffer_delay",
":jitter_buffer_delay_interface",
":jitter_buffer_delay_proxy",
":media_stream",
":rtp_receiver",
":video_rtp_track_source",
@ -556,6 +553,7 @@ rtc_library("video_rtp_receiver") {
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:threading",
"../rtc_base/system:no_unique_address",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
@ -615,19 +613,6 @@ rtc_library("video_track") {
]
}
rtc_source_set("jitter_buffer_delay_interface") {
sources = [ "jitter_buffer_delay_interface.h" ]
deps = [
"../media:rtc_media_base",
"../rtc_base:refcount",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("sdp_state_provider") {
sources = [ "sdp_state_provider.h" ]
deps = [
@ -636,35 +621,19 @@ rtc_source_set("sdp_state_provider") {
]
}
rtc_source_set("jitter_buffer_delay_proxy") {
sources = [ "jitter_buffer_delay_proxy.h" ]
deps = [
":jitter_buffer_delay_interface",
"../api:libjingle_peerconnection_api",
"../media:rtc_media_base",
]
}
rtc_library("jitter_buffer_delay") {
sources = [
"jitter_buffer_delay.cc",
"jitter_buffer_delay.h",
]
deps = [
":jitter_buffer_delay_interface",
"../api:sequence_checker",
"../media:rtc_media_base",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:safe_conversions",
"../rtc_base:safe_minmax",
"../rtc_base:threading",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
"../rtc_base/system:no_unique_address",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("remote_audio_source") {
@ -1024,7 +993,6 @@ if (rtc_include_tests && !build_with_chromium) {
":dtmf_sender",
":integration_test_helpers",
":jitter_buffer_delay",
":jitter_buffer_delay_interface",
":media_stream",
":peerconnection",
":remote_audio_source",
@ -1092,6 +1060,7 @@ if (rtc_include_tests && !build_with_chromium) {
"../test:field_trial",
"../test:fileutils",
"../test:rtp_test_utils",
"../test:test_common",
"../test/pc/sctp:fake_sctp_transport",
"./scenario_tests:pc_scenario_tests",
"//third_party/abseil-cpp/absl/algorithm:container",
@ -1181,7 +1150,6 @@ if (rtc_include_tests && !build_with_chromium) {
":audio_track",
":dtmf_sender",
":jitter_buffer_delay",
":jitter_buffer_delay_interface",
":media_stream",
":pc_test_utils",
":peerconnection",
@ -1291,7 +1259,6 @@ if (rtc_include_tests && !build_with_chromium) {
"test/frame_generator_capturer_video_track_source.h",
"test/mock_channel_interface.h",
"test/mock_data_channel.h",
"test/mock_delayable.h",
"test/mock_peer_connection_observers.h",
"test/mock_rtp_receiver_internal.h",
"test/mock_rtp_sender_internal.h",
@ -1303,7 +1270,6 @@ if (rtc_include_tests && !build_with_chromium) {
deps = [
":jitter_buffer_delay",
":jitter_buffer_delay_interface",
":libjingle_peerconnection",
":peerconnection",
":rtc_pc_base",

228
pc/audio_rtp_receiver.cc
View file

@ -18,11 +18,10 @@
#include "api/media_stream_track_proxy.h"
#include "api/sequence_checker.h"
#include "pc/audio_track.h"
#include "pc/jitter_buffer_delay.h"
#include "pc/jitter_buffer_delay_proxy.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/task_utils/to_queued_task.h"
namespace webrtc {
@ -52,10 +51,7 @@ AudioRtpReceiver::AudioRtpReceiver(
AudioTrack::Create(receiver_id, source_))),
cached_track_enabled_(track_->enabled()),
attachment_id_(GenerateUniqueId()),
delay_(JitterBufferDelayProxy::Create(
rtc::Thread::Current(),
worker_thread_,
rtc::make_ref_counted<JitterBufferDelay>(worker_thread))) {
worker_thread_safety_(PendingTaskSafetyFlag::CreateDetachedInactive()) {
RTC_DCHECK(worker_thread_);
RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
@ -64,140 +60,188 @@ AudioRtpReceiver::AudioRtpReceiver(
}
AudioRtpReceiver::~AudioRtpReceiver() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RTC_DCHECK(stopped_);
RTC_DCHECK(!media_channel_);
track_->GetSource()->UnregisterAudioObserver(this);
track_->UnregisterObserver(this);
Stop();
}
void AudioRtpReceiver::OnChanged() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
Reconfigure();
worker_thread_->PostTask(ToQueuedTask(
worker_thread_safety_,
[this, enabled = cached_track_enabled_, volume = cached_volume_]() {
RTC_DCHECK_RUN_ON(worker_thread_);
Reconfigure(enabled, volume);
}));
}
}
bool AudioRtpReceiver::SetOutputVolume(double volume) {
// RTC_RUN_ON(worker_thread_)
void AudioRtpReceiver::SetOutputVolume_w(double volume) {
RTC_DCHECK_GE(volume, 0.0);
RTC_DCHECK_LE(volume, 10.0);
RTC_DCHECK(media_channel_);
RTC_DCHECK(!stopped_);
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return ssrc_ ? media_channel_->SetOutputVolume(*ssrc_, volume)
ssrc_ ? media_channel_->SetOutputVolume(*ssrc_, volume)
: media_channel_->SetDefaultOutputVolume(volume);
});
}
void AudioRtpReceiver::OnSetVolume(double volume) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
cached_volume_ = volume;
if (!media_channel_ || stopped_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::OnSetVolume: No audio channel exists.";
if (stopped_)
return;
}
cached_volume_ = volume;
// When the track is disabled, the volume of the source, which is the
// corresponding WebRtc Voice Engine channel will be 0. So we do not allow
// setting the volume to the source when the track is disabled.
if (!stopped_ && track_->enabled()) {
if (!SetOutputVolume(cached_volume_)) {
RTC_NOTREACHED();
}
if (track_->enabled()) {
worker_thread_->PostTask(
ToQueuedTask(worker_thread_safety_, [this, volume = cached_volume_]() {
RTC_DCHECK_RUN_ON(worker_thread_);
SetOutputVolume_w(volume);
}));
}
}
rtc::scoped_refptr<DtlsTransportInterface> AudioRtpReceiver::dtls_transport()
const {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
return dtls_transport_;
}
std::vector<std::string> AudioRtpReceiver::stream_ids() const {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
std::vector<std::string> stream_ids(streams_.size());
for (size_t i = 0; i < streams_.size(); ++i)
stream_ids[i] = streams_[i]->id();
return stream_ids;
}
std::vector<rtc::scoped_refptr<MediaStreamInterface>>
AudioRtpReceiver::streams() const {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
return streams_;
}
RtpParameters AudioRtpReceiver::GetParameters() const {
if (!media_channel_ || stopped_) {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!media_channel_)
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_)
: media_channel_->GetDefaultRtpReceiveParameters();
});
}
void AudioRtpReceiver::SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
RTC_DCHECK_RUN_ON(worker_thread_);
frame_decryptor_ = std::move(frame_decryptor);
// Special Case: Set the frame decryptor to any value on any existing channel.
if (media_channel_ && ssrc_.has_value() && !stopped_) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
if (media_channel_ && ssrc_) {
media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
});
}
}
rtc::scoped_refptr<FrameDecryptorInterface>
AudioRtpReceiver::GetFrameDecryptor() const {
RTC_DCHECK_RUN_ON(worker_thread_);
return frame_decryptor_;
}
void AudioRtpReceiver::Stop() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
return;
}
if (!stopped_) {
source_->SetState(MediaSourceInterface::kEnded);
if (media_channel_) {
// Allow that SetOutputVolume fail. This is the normal case when the
// underlying media channel has already been deleted.
SetOutputVolume(0.0);
}
stopped_ = true;
}
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
RTC_DCHECK_RUN_ON(worker_thread_);
if (media_channel_)
SetOutputVolume_w(0.0);
SetMediaChannel_w(nullptr);
});
}
void AudioRtpReceiver::StopAndEndTrack() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
Stop();
track_->internal()->set_ended();
}
void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
RTC_DCHECK(media_channel_);
if (!stopped_ && ssrc_ == ssrc) {
return;
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
bool ok = worker_thread_->Invoke<bool>(
RTC_FROM_HERE, [&, enabled = cached_track_enabled_,
volume = cached_volume_, was_stopped = stopped_]() {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!media_channel_) {
RTC_DCHECK(was_stopped);
return false; // Can't restart.
}
if (!stopped_) {
source_->Stop(media_channel_, ssrc_);
delay_->OnStop();
if (!was_stopped && ssrc_ == ssrc) {
// Already running with that ssrc.
RTC_DCHECK(worker_thread_safety_->alive());
return true;
}
ssrc_ = ssrc;
if (!was_stopped) {
source_->Stop(media_channel_, ssrc_);
}
ssrc_ = std::move(ssrc);
source_->Start(media_channel_, ssrc_);
if (ssrc_) {
media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs());
}
Reconfigure(enabled, volume);
return true;
});
if (!ok)
return;
stopped_ = false;
source_->Start(media_channel_, ssrc);
delay_->OnStart(media_channel_, ssrc.value_or(0));
Reconfigure();
}
void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::SetupMediaChannel: No audio channel exists.";
return;
}
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RestartMediaChannel(ssrc);
}
void AudioRtpReceiver::SetupUnsignaledMediaChannel() {
if (!media_channel_) {
RTC_LOG(LS_ERROR) << "AudioRtpReceiver::SetupUnsignaledMediaChannel: No "
"audio channel exists.";
}
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RestartMediaChannel(absl::nullopt);
}
uint32_t AudioRtpReceiver::ssrc() const {
RTC_DCHECK_RUN_ON(worker_thread_);
return ssrc_.value_or(0);
}
void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
}
void AudioRtpReceiver::set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
dtls_transport_ = std::move(dtls_transport);
}
void AudioRtpReceiver::SetStreams(
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
// Remove remote track from any streams that are going away.
for (const auto& existing_stream : streams_) {
bool removed = true;
@ -230,51 +274,42 @@ void AudioRtpReceiver::SetStreams(
}
std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
if (!media_channel_ || !ssrc_ || stopped_) {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!media_channel_ || !ssrc_) {
return {};
}
return worker_thread_->Invoke<std::vector<RtpSource>>(
RTC_FROM_HERE, [&] { return media_channel_->GetSources(*ssrc_); });
return media_channel_->GetSources(*ssrc_);
}
void AudioRtpReceiver::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE, [this, frame_transformer = std::move(frame_transformer)] {
RTC_DCHECK_RUN_ON(worker_thread_);
frame_transformer_ = frame_transformer;
if (media_channel_ && ssrc_.has_value() && !stopped_) {
media_channel_->SetDepacketizerToDecoderFrameTransformer(
*ssrc_, frame_transformer);
if (media_channel_) {
media_channel_->SetDepacketizerToDecoderFrameTransformer(ssrc_.value_or(0),
frame_transformer);
}
});
frame_transformer_ = std::move(frame_transformer);
}
void AudioRtpReceiver::Reconfigure() {
if (!media_channel_ || stopped_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::Reconfigure: No audio channel exists.";
return;
}
if (!SetOutputVolume(track_->enabled() ? cached_volume_ : 0)) {
RTC_NOTREACHED();
}
// Reattach the frame decryptor if we were reconfigured.
MaybeAttachFrameDecryptorToMediaChannel(
ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
// RTC_RUN_ON(worker_thread_)
void AudioRtpReceiver::Reconfigure(bool track_enabled, double volume) {
RTC_DCHECK(media_channel_);
if (media_channel_ && ssrc_.has_value() && !stopped_) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!frame_transformer_)
return;
SetOutputVolume_w(track_enabled ? volume : 0);
if (ssrc_ && frame_decryptor_) {
// Reattach the frame decryptor if we were reconfigured.
media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
}
if (frame_transformer_) {
media_channel_->SetDepacketizerToDecoderFrameTransformer(
*ssrc_, frame_transformer_);
});
ssrc_.value_or(0), frame_transformer_);
}
}
void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
observer_ = observer;
// Deliver any notifications the observer may have missed by being set late.
if (received_first_packet_ && observer_) {
@ -284,16 +319,35 @@ void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
void AudioRtpReceiver::SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds) {
delay_->Set(delay_seconds);
RTC_DCHECK_RUN_ON(worker_thread_);
delay_.Set(delay_seconds);
if (media_channel_ && ssrc_)
media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs());
}
void AudioRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RTC_DCHECK(media_channel == nullptr ||
media_channel->media_type() == media_type());
if (stopped_ && !media_channel)
return;
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(worker_thread_);
SetMediaChannel_w(media_channel);
});
}
// RTC_RUN_ON(worker_thread_)
void AudioRtpReceiver::SetMediaChannel_w(cricket::MediaChannel* media_channel) {
media_channel ? worker_thread_safety_->SetAlive()
: worker_thread_safety_->SetNotAlive();
media_channel_ = static_cast<cricket::VoiceMediaChannel*>(media_channel);
}
void AudioRtpReceiver::NotifyFirstPacketReceived() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}

69
pc/audio_rtp_receiver.h
View file

@ -12,6 +12,7 @@
#define PC_AUDIO_RTP_RECEIVER_H_
#include <stdint.h>
#include <string>
#include <vector>
@ -25,13 +26,16 @@
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/transport/rtp/rtp_source.h"
#include "media/base/media_channel.h"
#include "pc/audio_track.h"
#include "pc/jitter_buffer_delay_interface.h"
#include "pc/jitter_buffer_delay.h"
#include "pc/remote_audio_source.h"
#include "pc/rtp_receiver.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
@ -39,7 +43,7 @@ namespace webrtc {
class AudioRtpReceiver : public ObserverInterface,
public AudioSourceInterface::AudioObserver,
public rtc::RefCountedObject<RtpReceiverInternal> {
public RtpReceiverInternal {
public:
AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
@ -59,22 +63,16 @@ class AudioRtpReceiver : public ObserverInterface,
// AudioSourceInterface::AudioObserver implementation
void OnSetVolume(double volume) override;
rtc::scoped_refptr<AudioTrackInterface> audio_track() const {
return track_.get();
}
rtc::scoped_refptr<AudioTrackInterface> audio_track() const { return track_; }
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
return dtls_transport_;
return track_;
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override;
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
return streams_;
}
const override;
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
@ -95,13 +93,11 @@ class AudioRtpReceiver : public ObserverInterface,
void StopAndEndTrack() override;
void SetupMediaChannel(uint32_t ssrc) override;
void SetupUnsignaledMediaChannel() override;
uint32_t ssrc() const override { return ssrc_.value_or(0); }
uint32_t ssrc() const override;
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
dtls_transport_ = dtls_transport;
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override;
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
@ -119,29 +115,40 @@ class AudioRtpReceiver : public ObserverInterface,
private:
void RestartMediaChannel(absl::optional<uint32_t> ssrc);
void Reconfigure();
bool SetOutputVolume(double volume);
void Reconfigure(bool track_enabled, double volume)
RTC_RUN_ON(worker_thread_);
void SetOutputVolume_w(double volume) RTC_RUN_ON(worker_thread_);
void SetMediaChannel_w(cricket::MediaChannel* media_channel)
RTC_RUN_ON(worker_thread_);
RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
rtc::Thread* const worker_thread_;
const std::string id_;
const rtc::scoped_refptr<RemoteAudioSource> source_;
const rtc::scoped_refptr<AudioTrackProxyWithInternal<AudioTrack>> track_;
cricket::VoiceMediaChannel* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool cached_track_enabled_;
double cached_volume_ = 1;
bool stopped_ = true;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
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
cricket::VoiceMediaChannel* media_channel_ RTC_GUARDED_BY(worker_thread_) =
nullptr;
absl::optional<uint32_t> ssrc_ RTC_GUARDED_BY(worker_thread_);
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_
RTC_GUARDED_BY(&signaling_thread_checker_);
bool cached_track_enabled_ RTC_GUARDED_BY(&signaling_thread_checker_);
double cached_volume_ RTC_GUARDED_BY(&signaling_thread_checker_) = 1.0;
bool stopped_ RTC_GUARDED_BY(&signaling_thread_checker_) = true;
RtpReceiverObserverInterface* observer_
RTC_GUARDED_BY(&signaling_thread_checker_) = nullptr;
bool received_first_packet_ RTC_GUARDED_BY(&signaling_thread_checker_) =
false;
const int attachment_id_;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_
RTC_GUARDED_BY(worker_thread_);
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_
RTC_GUARDED_BY(&signaling_thread_checker_);
// Stores and updates the playout delay. Handles caching cases if
// |SetJitterBufferMinimumDelay| is called before start.
rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
JitterBufferDelay delay_ RTC_GUARDED_BY(worker_thread_);
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
RTC_GUARDED_BY(worker_thread_);
const rtc::scoped_refptr<PendingTaskSafetyFlag> worker_thread_safety_;
};
} // namespace webrtc

45
pc/jitter_buffer_delay.cc
View file

@ -14,7 +14,6 @@
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/thread.h"
namespace {
constexpr int kDefaultDelay = 0;
@ -23,43 +22,21 @@ constexpr int kMaximumDelayMs = 10000;
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;
JitterBufferDelay::JitterBufferDelay() {
worker_thread_checker_.Detach();
}
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);
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
cached_delay_seconds_ = delay_seconds;
if (media_channel_ && ssrc_) {
media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms);
}
}
int JitterBufferDelay::GetMs() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return rtc::SafeClamp(
rtc::saturated_cast<int>(cached_delay_seconds_.value_or(kDefaultDelay) *
1000),
0, kMaximumDelayMs);
}
} // namespace webrtc

31
pc/jitter_buffer_delay.h
View file

@ -14,36 +14,25 @@
#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"
#include "api/sequence_checker.h"
#include "rtc_base/system/no_unique_address.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 {
// the start of media_channel by caching its request.
class JitterBufferDelay {
public:
// Must be called on signaling thread.
explicit JitterBufferDelay(rtc::Thread* worker_thread);
JitterBufferDelay();
void OnStart(cricket::Delayable* media_channel, uint32_t ssrc) override;
void OnStop() override;
void Set(absl::optional<double> delay_seconds) override;
void Set(absl::optional<double> delay_seconds);
int GetMs() const;
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_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
absl::optional<double> cached_delay_seconds_
RTC_GUARDED_BY(&worker_thread_checker_);
};
} // namespace webrtc

39
pc/jitter_buffer_delay_interface.h
View file

@ -1,39 +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_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_

31
pc/jitter_buffer_delay_proxy.h
View file

@ -1,31 +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_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/jitter_buffer_delay_interface.h"
namespace webrtc {
BEGIN_PROXY_MAP(JitterBufferDelay)
PROXY_PRIMARY_THREAD_DESTRUCTOR()
PROXY_METHOD2(void, OnStart, cricket::Delayable*, uint32_t)
PROXY_METHOD0(void, OnStop)
PROXY_SECONDARY_METHOD1(void, Set, absl::optional<double>)
END_PROXY_MAP()
} // namespace webrtc
#endif // PC_JITTER_BUFFER_DELAY_PROXY_H_

64
pc/jitter_buffer_delay_unittest.cc
View file

@ -13,79 +13,47 @@
#include <stdint.h>
#include "absl/types/optional.h"
#include "api/scoped_refptr.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_(
rtc::make_ref_counted<JitterBufferDelay>(rtc::Thread::Current())) {}
JitterBufferDelayTest() {}
protected:
rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
MockDelayable delayable_;
JitterBufferDelay delay_;
};
TEST_F(JitterBufferDelayTest, Set) {
delay_->OnStart(&delayable_, kSsrc);
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 3000))
.WillOnce(Return(true));
// Delay in seconds.
delay_->Set(3.0);
delay_.Set(3.0);
EXPECT_EQ(delay_.GetMs(), 3000);
}
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, DefaultValue) {
EXPECT_EQ(delay_.GetMs(), 0); // Default value is 0ms.
}
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);
delay_.Set(10.5);
EXPECT_EQ(delay_.GetMs(), 10000);
// Test int overflow.
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 10000))
.WillOnce(Return(true));
delay_->Set(21474836470.0);
delay_.Set(21474836470.0);
EXPECT_EQ(delay_.GetMs(), 10000);
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
.WillOnce(Return(true));
delay_->Set(-21474836470.0);
delay_.Set(-21474836470.0);
EXPECT_EQ(delay_.GetMs(), 0);
// Boundary value in seconds to milliseconds conversion.
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
.WillOnce(Return(true));
delay_->Set(0.0009);
delay_.Set(0.0009);
EXPECT_EQ(delay_.GetMs(), 0);
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
.WillOnce(Return(true));
delay_->Set(-2.0);
delay_.Set(-2.0);
EXPECT_EQ(delay_.GetMs(), 0);
}
} // namespace webrtc

28
pc/remote_audio_source.cc
View file

@ -61,7 +61,7 @@ RemoteAudioSource::RemoteAudioSource(
}
RemoteAudioSource::~RemoteAudioSource() {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(audio_observers_.empty());
if (!sinks_.empty()) {
RTC_LOG(LS_WARNING)
@ -71,32 +71,28 @@ RemoteAudioSource::~RemoteAudioSource() {
void RemoteAudioSource::Start(cricket::VoiceMediaChannel* media_channel,
absl::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
RTC_DCHECK_RUN_ON(worker_thread_);
// Register for callbacks immediately before AddSink so that we always get
// notified when a channel goes out of scope (signaled when "AudioDataProxy"
// is destroyed).
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
ssrc ? media_channel->SetRawAudioSink(
*ssrc, std::make_unique<AudioDataProxy>(this))
RTC_DCHECK(media_channel);
ssrc ? media_channel->SetRawAudioSink(*ssrc,
std::make_unique<AudioDataProxy>(this))
: media_channel->SetDefaultRawAudioSink(
std::make_unique<AudioDataProxy>(this));
});
}
void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel,
absl::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(media_channel);
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr)
: media_channel->SetDefaultRawAudioSink(nullptr);
});
}
void RemoteAudioSource::SetState(SourceState new_state) {
RTC_DCHECK_RUN_ON(main_thread_);
if (state_ != new_state) {
state_ = new_state;
FireOnChanged();
@ -104,12 +100,12 @@ void RemoteAudioSource::SetState(SourceState new_state) {
}
MediaSourceInterface::SourceState RemoteAudioSource::state() const {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK_RUN_ON(main_thread_);
return state_;
}
bool RemoteAudioSource::remote() const {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK_RUN_ON(main_thread_);
return true;
}
@ -135,7 +131,7 @@ void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
}
void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(sink);
if (state_ != MediaSourceInterface::kLive) {
@ -149,7 +145,7 @@ void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
}
void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(sink);
MutexLock lock(&sink_lock_);
@ -184,7 +180,7 @@ void RemoteAudioSource::OnAudioChannelGone() {
}
void RemoteAudioSource::OnMessage(rtc::Message* msg) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK_RUN_ON(main_thread_);
sinks_.clear();
SetState(MediaSourceInterface::kEnded);
// Will possibly delete this RemoteAudioSource since it is reference counted

16
pc/rtp_receiver.cc
View file

@ -39,20 +39,4 @@ RtpReceiverInternal::CreateStreamsFromIds(std::vector<std::string> stream_ids) {
return streams;
}
// Attempt to attach the frame decryptor to the current media channel on the
// correct worker thread only if both the media channel exists and a ssrc has
// been allocated to the stream.
void RtpReceiverInternal::MaybeAttachFrameDecryptorToMediaChannel(
const absl::optional<uint32_t>& ssrc,
rtc::Thread* worker_thread,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
cricket::MediaChannel* media_channel,
bool stopped) {
if (media_channel && frame_decryptor && ssrc.has_value() && !stopped) {
worker_thread->Invoke<void>(RTC_FROM_HERE, [&] {
media_channel->SetFrameDecryptor(*ssrc, frame_decryptor);
});
}
}
} // namespace webrtc

7
pc/rtp_receiver.h
View file

@ -92,13 +92,6 @@ class RtpReceiverInternal : public RtpReceiverInterface {
static std::vector<rtc::scoped_refptr<MediaStreamInterface>>
CreateStreamsFromIds(std::vector<std::string> stream_ids);
static void MaybeAttachFrameDecryptorToMediaChannel(
const absl::optional<uint32_t>& ssrc,
rtc::Thread* worker_thread,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
cricket::MediaChannel* media_channel,
bool stopped);
};
} // namespace webrtc

36
pc/rtp_sender_receiver_unittest.cc
View file

@ -63,6 +63,7 @@
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/run_loop.h"
using ::testing::_;
using ::testing::ContainerEq;
@ -299,8 +300,8 @@ class RtpSenderReceiverTest
void CreateAudioRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
audio_rtp_receiver_ =
new AudioRtpReceiver(rtc::Thread::Current(), kAudioTrackId, streams,
audio_rtp_receiver_ = rtc::make_ref_counted<AudioRtpReceiver>(
rtc::Thread::Current(), kAudioTrackId, streams,
/*is_unified_plan=*/true);
audio_rtp_receiver_->SetMediaChannel(voice_media_channel_);
audio_rtp_receiver_->SetupMediaChannel(kAudioSsrc);
@ -310,8 +311,8 @@ class RtpSenderReceiverTest
void CreateVideoRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
video_rtp_receiver_ =
new VideoRtpReceiver(rtc::Thread::Current(), kVideoTrackId, streams);
video_rtp_receiver_ = rtc::make_ref_counted<VideoRtpReceiver>(
rtc::Thread::Current(), kVideoTrackId, streams);
video_rtp_receiver_->SetMediaChannel(video_media_channel_);
video_rtp_receiver_->SetupMediaChannel(kVideoSsrc);
video_track_ = video_rtp_receiver_->video_track();
@ -330,19 +331,25 @@ class RtpSenderReceiverTest
video_media_channel_->AddRecvStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
video_rtp_receiver_ =
new VideoRtpReceiver(rtc::Thread::Current(), kVideoTrackId, streams);
video_rtp_receiver_ = rtc::make_ref_counted<VideoRtpReceiver>(
rtc::Thread::Current(), kVideoTrackId, streams);
video_rtp_receiver_->SetMediaChannel(video_media_channel_);
video_rtp_receiver_->SetupMediaChannel(primary_ssrc);
video_track_ = video_rtp_receiver_->video_track();
}
void DestroyAudioRtpReceiver() {
if (!audio_rtp_receiver_)
return;
audio_rtp_receiver_->Stop();
audio_rtp_receiver_ = nullptr;
VerifyVoiceChannelNoOutput();
}
void DestroyVideoRtpReceiver() {
if (!video_rtp_receiver_)
return;
video_rtp_receiver_->Stop();
video_rtp_receiver_ = nullptr;
VerifyVideoChannelNoOutput();
}
@ -498,6 +505,7 @@ class RtpSenderReceiverTest
}
protected:
test::RunLoop run_loop_;
rtc::Thread* const network_thread_;
rtc::Thread* const worker_thread_;
webrtc::RtcEventLogNull event_log_;
@ -599,11 +607,15 @@ TEST_F(RtpSenderReceiverTest, RemoteAudioTrackDisable) {
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(1, volume);
// Handling of enable/disable is applied asynchronously.
audio_track_->set_enabled(false);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0, volume);
audio_track_->set_enabled(true);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(1, volume);
@ -636,6 +648,7 @@ TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) {
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state());
EXPECT_EQ(webrtc::MediaSourceInterface::kEnded,
video_track_->GetSource()->state());
DestroyVideoRtpReceiver();
}
// Currently no action is taken when a remote video track is disabled or
@ -657,22 +670,27 @@ TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetVolume) {
double volume;
audio_track_->GetSource()->SetVolume(0.5);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0.5, volume);
// Disable the audio track, this should prevent setting the volume.
audio_track_->set_enabled(false);
RTC_DCHECK_EQ(worker_thread_, run_loop_.task_queue());
run_loop_.Flush();
audio_track_->GetSource()->SetVolume(0.8);
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0, volume);
// When the track is enabled, the previously set volume should take effect.
audio_track_->set_enabled(true);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0.8, volume);
// Try changing volume one more time.
audio_track_->GetSource()->SetVolume(0.9);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0.9, volume);
@ -683,12 +701,14 @@ TEST_F(RtpSenderReceiverTest, AudioRtpReceiverDelay) {
CreateAudioRtpReceiver();
VerifyRtpReceiverDelayBehaviour(voice_media_channel_,
audio_rtp_receiver_.get(), kAudioSsrc);
DestroyAudioRtpReceiver();
}
TEST_F(RtpSenderReceiverTest, VideoRtpReceiverDelay) {
CreateVideoRtpReceiver();
VerifyRtpReceiverDelayBehaviour(video_media_channel_,
video_rtp_receiver_.get(), kVideoSsrc);
DestroyVideoRtpReceiver();
}
// Test that the media channel isn't enabled for sending if the audio sender
@ -1582,6 +1602,7 @@ TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetFrameDecryptor) {
audio_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
EXPECT_EQ(fake_frame_decryptor.get(),
audio_rtp_receiver_->GetFrameDecryptor().get());
DestroyAudioRtpReceiver();
}
// Validate that the default FrameEncryptor setting is nullptr.
@ -1593,6 +1614,7 @@ TEST_F(RtpSenderReceiverTest, AudioReceiverCannotSetFrameDecryptorAfterStop) {
audio_rtp_receiver_->Stop();
audio_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
// TODO(webrtc:9926) - Validate media channel not set once fakes updated.
DestroyAudioRtpReceiver();
}
// Validate that the default FrameEncryptor setting is nullptr.
@ -1627,6 +1649,7 @@ TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetFrameDecryptor) {
video_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
EXPECT_EQ(fake_frame_decryptor.get(),
video_rtp_receiver_->GetFrameDecryptor().get());
DestroyVideoRtpReceiver();
}
// Validate that the default FrameEncryptor setting is nullptr.
@ -1638,6 +1661,7 @@ TEST_F(RtpSenderReceiverTest, VideoReceiverCannotSetFrameDecryptorAfterStop) {
video_rtp_receiver_->Stop();
video_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
// TODO(webrtc:9926) - Validate media channel not set once fakes updated.
DestroyVideoRtpReceiver();
}
// Checks that calling the internal methods for get/set parameters do not

14
pc/rtp_transceiver.cc
View file

@ -211,14 +211,10 @@ void RtpTransceiver::SetChannel(cricket::ChannelInterface* channel) {
for (const auto& receiver : receivers_) {
if (!channel_) {
// TODO(tommi): This can internally block and hop to the worker thread.
// It's likely that SetMediaChannel also does that, so perhaps we should
// require SetMediaChannel(nullptr) to also Stop() and skip this call.
receiver->internal()->Stop();
} else {
receiver->internal()->SetMediaChannel(channel_->media_channel());
}
receiver->internal()->SetMediaChannel(channel_ ? channel_->media_channel()
: nullptr);
}
}
@ -268,12 +264,8 @@ bool RtpTransceiver::RemoveReceiver(RtpReceiverInterface* receiver) {
if (it == receivers_.end()) {
return false;
}
// `Stop()` will clear the internally cached pointer to the media channel.
(*it)->internal()->Stop();
// After the receiver has been removed, there's no guarantee that the
// contained media channel isn't deleted shortly after this. To make sure that
// the receiver doesn't spontaneously try to use it's (potentially stale)
// media channel reference, we clear it out.
(*it)->internal()->SetMediaChannel(nullptr);
receivers_.erase(it);
return true;
}

2
pc/rtp_transceiver_unittest.cc
View file

@ -93,6 +93,7 @@ class RtpTransceiverUnifiedPlanTest : public ::testing::Test {
rtc::Thread::Current(),
sender_),
RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
rtc::Thread::Current(),
rtc::Thread::Current(),
receiver_),
channel_manager_.get(),
@ -162,6 +163,7 @@ class RtpTransceiverTestForHeaderExtensions : public ::testing::Test {
rtc::Thread::Current(),
sender_),
RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
rtc::Thread::Current(),
rtc::Thread::Current(),
receiver_),
channel_manager_.get(),

19
pc/rtp_transmission_manager.cc
View file

@ -11,6 +11,7 @@
#include "pc/rtp_transmission_manager.h"
#include <algorithm>
#include <utility>
#include "absl/types/optional.h"
#include "api/peer_connection_interface.h"
@ -240,14 +241,16 @@ RtpTransmissionManager::CreateReceiver(cricket::MediaType media_type,
receiver;
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
new AudioRtpReceiver(worker_thread(), receiver_id,
std::vector<std::string>({}), IsUnifiedPlan()));
signaling_thread(), worker_thread(),
rtc::make_ref_counted<AudioRtpReceiver>(worker_thread(), receiver_id,
std::vector<std::string>({}),
IsUnifiedPlan()));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), new VideoRtpReceiver(worker_thread(), receiver_id,
signaling_thread(), worker_thread(),
rtc::make_ref_counted<VideoRtpReceiver>(worker_thread(), receiver_id,
std::vector<std::string>({})));
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
}
@ -453,7 +456,7 @@ void RtpTransmissionManager::CreateAudioReceiver(
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
// the constructor taking stream IDs instead.
auto* audio_receiver = new AudioRtpReceiver(
auto audio_receiver = rtc::make_ref_counted<AudioRtpReceiver>(
worker_thread(), remote_sender_info.sender_id, streams, IsUnifiedPlan());
audio_receiver->SetMediaChannel(voice_media_channel());
if (remote_sender_info.sender_id == kDefaultAudioSenderId) {
@ -462,7 +465,7 @@ void RtpTransmissionManager::CreateAudioReceiver(
audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
}
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), audio_receiver);
signaling_thread(), worker_thread(), std::move(audio_receiver));
GetAudioTransceiver()->internal()->AddReceiver(receiver);
Observer()->OnAddTrack(receiver, streams);
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
@ -476,7 +479,7 @@ void RtpTransmissionManager::CreateVideoReceiver(
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
// the constructor taking stream IDs instead.
auto* video_receiver = new VideoRtpReceiver(
auto video_receiver = rtc::make_ref_counted<VideoRtpReceiver>(
worker_thread(), remote_sender_info.sender_id, streams);
video_receiver->SetMediaChannel(video_media_channel());
if (remote_sender_info.sender_id == kDefaultVideoSenderId) {
@ -485,7 +488,7 @@ void RtpTransmissionManager::CreateVideoReceiver(
video_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
}
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), video_receiver);
signaling_thread(), worker_thread(), std::move(video_receiver));
GetVideoTransceiver()->internal()->AddReceiver(receiver);
Observer()->OnAddTrack(receiver, streams);
NoteUsageEvent(UsageEvent::VIDEO_ADDED);

30
pc/stats_collector.cc
View file

@ -1163,9 +1163,10 @@ void StatsCollector::ExtractMediaInfo(
std::vector<std::unique_ptr<MediaChannelStatsGatherer>> gatherers;
auto transceivers = pc_->GetTransceiversInternal();
{
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const auto& transceiver : pc_->GetTransceiversInternal()) {
for (const auto& transceiver : transceivers) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (!channel) {
continue;
@ -1176,20 +1177,37 @@ void StatsCollector::ExtractMediaInfo(
gatherer->transport_name = transport_names_by_mid.at(gatherer->mid);
for (const auto& sender : transceiver->internal()->senders()) {
std::string track_id = (sender->track() ? sender->track()->id() : "");
auto track = sender->track();
std::string track_id = (track ? track->id() : "");
gatherer->sender_track_id_by_ssrc.insert(
std::make_pair(sender->ssrc(), track_id));
}
for (const auto& receiver : transceiver->internal()->receivers()) {
gatherer->receiver_track_id_by_ssrc.insert(std::make_pair(
receiver->internal()->ssrc(), receiver->track()->id()));
}
// Populating `receiver_track_id_by_ssrc` will be done on the worker
// thread as the `ssrc` property of the receiver needs to be accessed
// there.
gatherers.push_back(std::move(gatherer));
}
}
pc_->worker_thread()->Invoke<void>(RTC_FROM_HERE, [&] {
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
// Populate `receiver_track_id_by_ssrc` for the gatherers.
int i = 0;
for (const auto& transceiver : transceivers) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (!channel)
continue;
MediaChannelStatsGatherer* gatherer = gatherers[i++].get();
RTC_DCHECK_EQ(gatherer->mid, channel->content_name());
for (const auto& receiver : transceiver->internal()->receivers()) {
gatherer->receiver_track_id_by_ssrc.insert(std::make_pair(
receiver->internal()->ssrc(), receiver->track()->id()));
}
}
for (auto it = gatherers.begin(); it != gatherers.end();
/* incremented manually */) {
MediaChannelStatsGatherer* gatherer = it->get();

2
pc/test/fake_peer_connection_for_stats.h
View file

@ -182,7 +182,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
// TODO(steveanton): Switch tests to use RtpTransceivers directly.
auto receiver_proxy =
RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread_, receiver);
signaling_thread_, worker_thread_, receiver);
GetOrCreateFirstTransceiverOfType(receiver->media_type())
->internal()
->AddReceiver(receiver_proxy);

202
pc/video_rtp_receiver.cc
View file

@ -17,8 +17,6 @@
#include "api/video/recordable_encoded_frame.h"
#include "api/video_track_source_proxy.h"
#include "pc/jitter_buffer_delay.h"
#include "pc/jitter_buffer_delay_proxy.h"
#include "pc/video_track.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
@ -39,7 +37,7 @@ VideoRtpReceiver::VideoRtpReceiver(
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams)
: worker_thread_(worker_thread),
id_(receiver_id),
source_(new RefCountedObject<VideoRtpTrackSource>(this)),
source_(rtc::make_ref_counted<VideoRtpTrackSource>(&source_callback_)),
track_(VideoTrackProxyWithInternal<VideoTrack>::Create(
rtc::Thread::Current(),
worker_thread,
@ -49,111 +47,130 @@ VideoRtpReceiver::VideoRtpReceiver(
worker_thread,
source_),
worker_thread))),
attachment_id_(GenerateUniqueId()),
delay_(JitterBufferDelayProxy::Create(
rtc::Thread::Current(),
worker_thread,
new rtc::RefCountedObject<JitterBufferDelay>(worker_thread))) {
attachment_id_(GenerateUniqueId()) {
RTC_DCHECK(worker_thread_);
SetStreams(streams);
source_->SetState(MediaSourceInterface::kLive);
}
VideoRtpReceiver::~VideoRtpReceiver() {
// Since cricket::VideoRenderer is not reference counted,
// we need to remove it from the channel before we are deleted.
Stop();
// Make sure we can't be called by the |source_| anymore.
worker_thread_->Invoke<void>(RTC_FROM_HERE,
[this] { source_->ClearCallback(); });
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RTC_DCHECK(stopped_);
RTC_DCHECK(!media_channel_);
}
std::vector<std::string> VideoRtpReceiver::stream_ids() const {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
std::vector<std::string> stream_ids(streams_.size());
for (size_t i = 0; i < streams_.size(); ++i)
stream_ids[i] = streams_[i]->id();
return stream_ids;
}
rtc::scoped_refptr<DtlsTransportInterface> VideoRtpReceiver::dtls_transport()
const {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
return dtls_transport_;
}
std::vector<rtc::scoped_refptr<MediaStreamInterface>>
VideoRtpReceiver::streams() const {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
return streams_;
}
RtpParameters VideoRtpReceiver::GetParameters() const {
if (!media_channel_ || stopped_) {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!media_channel_)
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_)
: media_channel_->GetDefaultRtpReceiveParameters();
});
}
void VideoRtpReceiver::SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
RTC_DCHECK_RUN_ON(worker_thread_);
frame_decryptor_ = std::move(frame_decryptor);
// Special Case: Set the frame decryptor to any value on any existing channel.
if (media_channel_ && ssrc_.has_value() && !stopped_) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
if (media_channel_ && ssrc_) {
media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
});
}
}
rtc::scoped_refptr<FrameDecryptorInterface>
VideoRtpReceiver::GetFrameDecryptor() const {
RTC_DCHECK_RUN_ON(worker_thread_);
return frame_decryptor_;
}
void VideoRtpReceiver::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(worker_thread_);
frame_transformer_ = std::move(frame_transformer);
if (media_channel_ && !stopped_) {
if (media_channel_) {
media_channel_->SetDepacketizerToDecoderFrameTransformer(
ssrc_.value_or(0), frame_transformer_);
}
});
}
void VideoRtpReceiver::Stop() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
return;
}
if (!stopped_) {
source_->SetState(MediaSourceInterface::kEnded);
if (!media_channel_) {
RTC_LOG(LS_WARNING) << "VideoRtpReceiver::Stop: No video channel exists.";
} else {
// Allow that SetSink fails. This is the normal case when the underlying
// media channel has already been deleted.
stopped_ = true;
}
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(worker_thread_);
if (media_channel_) {
SetSink(nullptr);
});
SetMediaChannel_w(nullptr);
}
delay_->OnStop();
stopped_ = true;
source_->ClearCallback();
});
}
void VideoRtpReceiver::StopAndEndTrack() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
Stop();
track_->internal()->set_ended();
}
void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
RTC_DCHECK(media_channel_);
if (!stopped_ && ssrc_ == ssrc) {
return;
}
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
// `stopped_` will be `true` on construction. RestartMediaChannel
// can in this case function like "ensure started" and flip `stopped_`
// to false.
// TODO(tommi): Can we restart the media channel without blocking?
bool ok = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&, was_stopped =
stopped_] {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!stopped_) {
if (!media_channel_) {
// Ignore further negotiations if we've already been stopped and don't
// have an associated media channel.
RTC_DCHECK(was_stopped);
return false; // Can't restart.
}
if (!was_stopped && ssrc_ == ssrc) {
// Already running with that ssrc.
return true;
}
// Disconnect from the previous ssrc.
if (!was_stopped) {
SetSink(nullptr);
}
bool encoded_sink_enabled = saved_encoded_sink_enabled_;
SetEncodedSinkEnabled(false);
stopped_ = false;
ssrc_ = ssrc;
// Set up the new ssrc.
ssrc_ = std::move(ssrc);
SetSink(source_->sink());
if (encoded_sink_enabled) {
SetEncodedSinkEnabled(true);
@ -163,47 +180,62 @@ void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
media_channel_->SetDepacketizerToDecoderFrameTransformer(
ssrc_.value_or(0), frame_transformer_);
}
if (media_channel_ && ssrc_) {
if (frame_decryptor_) {
media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
}
media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs());
}
return true;
});
// Attach any existing frame decryptor to the media channel.
MaybeAttachFrameDecryptorToMediaChannel(
ssrc, worker_thread_, frame_decryptor_, media_channel_, stopped_);
// TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC
// value.
delay_->OnStart(media_channel_, ssrc.value_or(0));
if (!ok)
return;
stopped_ = false;
}
// RTC_RUN_ON(worker_thread_)
void VideoRtpReceiver::SetSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
RTC_DCHECK(media_channel_);
if (ssrc_) {
media_channel_->SetSink(*ssrc_, sink);
return;
}
} else {
media_channel_->SetDefaultSink(sink);
}
}
void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "VideoRtpReceiver::SetupMediaChannel: No video channel exists.";
}
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RestartMediaChannel(ssrc);
}
void VideoRtpReceiver::SetupUnsignaledMediaChannel() {
if (!media_channel_) {
RTC_LOG(LS_ERROR) << "VideoRtpReceiver::SetupUnsignaledMediaChannel: No "
"video channel exists.";
}
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RestartMediaChannel(absl::nullopt);
}
uint32_t VideoRtpReceiver::ssrc() const {
RTC_DCHECK_RUN_ON(worker_thread_);
return ssrc_.value_or(0);
}
void VideoRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
}
void VideoRtpReceiver::set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
dtls_transport_ = std::move(dtls_transport);
}
void VideoRtpReceiver::SetStreams(
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
// Remove remote track from any streams that are going away.
for (const auto& existing_stream : streams_) {
bool removed = true;
@ -236,6 +268,7 @@ void VideoRtpReceiver::SetStreams(
}
void VideoRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
observer_ = observer;
// Deliver any notifications the observer may have missed by being set late.
if (received_first_packet_ && observer_) {
@ -245,14 +278,31 @@ void VideoRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
void VideoRtpReceiver::SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds) {
delay_->Set(delay_seconds);
RTC_DCHECK_RUN_ON(worker_thread_);
delay_.Set(delay_seconds);
if (media_channel_ && ssrc_)
media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs());
}
void VideoRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
RTC_DCHECK(media_channel == nullptr ||
media_channel->media_type() == media_type());
if (stopped_ && !media_channel)
return;
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(worker_thread_);
SetMediaChannel_w(media_channel);
});
}
// RTC_RUN_ON(worker_thread_)
void VideoRtpReceiver::SetMediaChannel_w(cricket::MediaChannel* media_channel) {
if (media_channel == media_channel_)
return;
bool encoded_sink_enabled = saved_encoded_sink_enabled_;
if (encoded_sink_enabled && media_channel_) {
// Turn off the old sink, if any.
@ -275,10 +325,10 @@ void VideoRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
ssrc_.value_or(0), frame_transformer_);
}
}
});
}
void VideoRtpReceiver::NotifyFirstPacketReceived() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
@ -286,11 +336,10 @@ void VideoRtpReceiver::NotifyFirstPacketReceived() {
}
std::vector<RtpSource> VideoRtpReceiver::GetSources() const {
if (!media_channel_ || !ssrc_ || stopped_) {
return {};
}
return worker_thread_->Invoke<std::vector<RtpSource>>(
RTC_FROM_HERE, [&] { return media_channel_->GetSources(*ssrc_); });
RTC_DCHECK_RUN_ON(worker_thread_);
if (!ssrc_ || !media_channel_)
return std::vector<RtpSource>();
return media_channel_->GetSources(*ssrc_);
}
void VideoRtpReceiver::OnGenerateKeyFrame() {
@ -316,20 +365,21 @@ void VideoRtpReceiver::OnEncodedSinkEnabled(bool enable) {
saved_encoded_sink_enabled_ = enable;
}
// RTC_RUN_ON(worker_thread_)
void VideoRtpReceiver::SetEncodedSinkEnabled(bool enable) {
if (media_channel_) {
if (enable) {
if (!media_channel_)
return;
// TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC
auto source = source_;
const auto ssrc = ssrc_.value_or(0);
if (enable) {
media_channel_->SetRecordableEncodedFrameCallback(
ssrc_.value_or(0),
[source = std::move(source)](const RecordableEncodedFrame& frame) {
ssrc, [source = source_](const RecordableEncodedFrame& frame) {
source->BroadcastRecordableEncodedFrame(frame);
});
} else {
// TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC
media_channel_->ClearRecordableEncodedFrameCallback(ssrc_.value_or(0));
}
media_channel_->ClearRecordableEncodedFrameCallback(ssrc);
}
}

92
pc/video_rtp_receiver.h
View file

@ -32,18 +32,18 @@
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "media/base/media_channel.h"
#include "pc/jitter_buffer_delay_interface.h"
#include "pc/jitter_buffer_delay.h"
#include "pc/rtp_receiver.h"
#include "pc/video_rtp_track_source.h"
#include "pc/video_track.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
public VideoRtpTrackSource::Callback {
class VideoRtpReceiver : public RtpReceiverInternal {
public:
// An SSRC of 0 will create a receiver that will match the first SSRC it
// sees. Must be called on signaling thread.
@ -59,23 +59,16 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
virtual ~VideoRtpReceiver();
rtc::scoped_refptr<VideoTrackInterface> video_track() const {
return track_.get();
}
rtc::scoped_refptr<VideoTrackInterface> video_track() const { return track_; }
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
return dtls_transport_;
return track_;
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override;
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
return streams_;
}
const override;
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
@ -98,13 +91,11 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
void StopAndEndTrack() override;
void SetupMediaChannel(uint32_t ssrc) override;
void SetupUnsignaledMediaChannel() override;
uint32_t ssrc() const override { return ssrc_.value_or(0); }
uint32_t ssrc() const override;
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
dtls_transport_ = dtls_transport;
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override;
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
@ -123,33 +114,68 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
void RestartMediaChannel(absl::optional<uint32_t> ssrc);
void SetSink(rtc::VideoSinkInterface<VideoFrame>* sink)
RTC_RUN_ON(worker_thread_);
void SetMediaChannel_w(cricket::MediaChannel* media_channel)
RTC_RUN_ON(worker_thread_);
// VideoRtpTrackSource::Callback
void OnGenerateKeyFrame() override;
void OnEncodedSinkEnabled(bool enable) override;
void OnGenerateKeyFrame();
void OnEncodedSinkEnabled(bool enable);
void SetEncodedSinkEnabled(bool enable) RTC_RUN_ON(worker_thread_);
class SourceCallback : public VideoRtpTrackSource::Callback {
public:
explicit SourceCallback(VideoRtpReceiver* receiver) : receiver_(receiver) {}
~SourceCallback() override = default;
private:
void OnGenerateKeyFrame() override { receiver_->OnGenerateKeyFrame(); }
void OnEncodedSinkEnabled(bool enable) override {
receiver_->OnEncodedSinkEnabled(enable);
}
VideoRtpReceiver* const receiver_;
} source_callback_{this};
RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
rtc::Thread* const worker_thread_;
const std::string id_;
cricket::VideoMediaChannel* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
// See documentation for `stopped_` below for when a valid media channel
// has been assigned and when this pointer will be null.
cricket::VideoMediaChannel* media_channel_ RTC_GUARDED_BY(worker_thread_) =
nullptr;
absl::optional<uint32_t> ssrc_ RTC_GUARDED_BY(worker_thread_);
// |source_| is held here to be able to change the state of the source when
// the VideoRtpReceiver is stopped.
rtc::scoped_refptr<VideoRtpTrackSource> source_;
rtc::scoped_refptr<VideoTrackProxyWithInternal<VideoTrack>> track_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool stopped_ = true;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
const rtc::scoped_refptr<VideoRtpTrackSource> source_;
const rtc::scoped_refptr<VideoTrackProxyWithInternal<VideoTrack>> track_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_
RTC_GUARDED_BY(&signaling_thread_checker_);
// `stopped` is state that's used on the signaling thread to indicate whether
// a valid `media_channel_` has been assigned and configured. When an instance
// of VideoRtpReceiver is initially created, `stopped_` is true and will
// remain true until either `SetupMediaChannel` or
// `SetupUnsignaledMediaChannel` is called after assigning a media channel.
// After that, `stopped_` will remain false until `Stop()` is called.
// Note, for checking the state of the class on the worker thread,
// check `media_channel_` instead, as that's the main worker thread state.
bool stopped_ RTC_GUARDED_BY(&signaling_thread_checker_) = true;
RtpReceiverObserverInterface* observer_
RTC_GUARDED_BY(&signaling_thread_checker_) = nullptr;
bool received_first_packet_ RTC_GUARDED_BY(&signaling_thread_checker_) =
false;
const int attachment_id_;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_
RTC_GUARDED_BY(worker_thread_);
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_
RTC_GUARDED_BY(&signaling_thread_checker_);
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
RTC_GUARDED_BY(worker_thread_);
// Allows to thread safely change jitter buffer delay. Handles caching cases
// Stores the minimum jitter buffer delay. Handles caching cases
// if |SetJitterBufferMinimumDelay| is called before start.
rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
JitterBufferDelay delay_ RTC_GUARDED_BY(worker_thread_);
// Records if we should generate a keyframe when |media_channel_| gets set up
// or switched.
bool saved_generate_keyframe_ RTC_GUARDED_BY(worker_thread_) = false;

25
pc/video_rtp_receiver_unittest.cc
View file

@ -17,8 +17,10 @@
#include "test/gmock.h"
using ::testing::_;
using ::testing::AnyNumber;
using ::testing::InSequence;
using ::testing::Mock;
using ::testing::NiceMock;
using ::testing::SaveArg;
using ::testing::StrictMock;
@ -53,19 +55,26 @@ class VideoRtpReceiverTest : public testing::Test {
VideoRtpReceiverTest()
: worker_thread_(rtc::Thread::Create()),
channel_(nullptr, cricket::VideoOptions()),
receiver_(new VideoRtpReceiver(worker_thread_.get(),
"receiver",
{"stream"})) {
receiver_(rtc::make_ref_counted<VideoRtpReceiver>(
worker_thread_.get(),
std::string("receiver"),
std::vector<std::string>({"stream"}))) {
worker_thread_->Start();
receiver_->SetMediaChannel(&channel_);
}
~VideoRtpReceiverTest() override {
// Clear expectations that tests may have set up before calling Stop().
Mock::VerifyAndClearExpectations(&channel_);
receiver_->Stop();
}
webrtc::VideoTrackSourceInterface* Source() {
return receiver_->streams()[0]->FindVideoTrack("receiver")->GetSource();
}
std::unique_ptr<rtc::Thread> worker_thread_;
MockVideoMediaChannel channel_;
NiceMock<MockVideoMediaChannel> channel_;
rtc::scoped_refptr<VideoRtpReceiver> receiver_;
};
@ -98,6 +107,10 @@ TEST_F(VideoRtpReceiverTest,
// Switching to a new channel should now not cause calls to GenerateKeyFrame.
StrictMock<MockVideoMediaChannel> channel4(nullptr, cricket::VideoOptions());
receiver_->SetMediaChannel(&channel4);
// We must call Stop() here since the mock media channels live on the stack
// and `receiver_` still has a pointer to those objects.
receiver_->Stop();
}
TEST_F(VideoRtpReceiverTest, EnablesEncodedOutput) {
@ -131,6 +144,10 @@ TEST_F(VideoRtpReceiverTest, DisablesEnablesEncodedOutputOnChannelSwitch) {
Source()->RemoveEncodedSink(&sink);
StrictMock<MockVideoMediaChannel> channel3(nullptr, cricket::VideoOptions());
receiver_->SetMediaChannel(&channel3);
// We must call Stop() here since the mock media channels live on the stack
// and `receiver_` still has a pointer to those objects.
receiver_->Stop();
}
TEST_F(VideoRtpReceiverTest, BroadcastsEncodedFramesWhenEnabled) {