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Remove Mutex from BaseChannel.

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There's a bit of copy/pasted code in the channel code, which is
making moving network traffic consistently over to the network thread
a bit trickier than it needs to be, so I'm also updating variable
names used in Set[Local|Remote]Content_w to be more explicitly the same
and make it clear that the code is copy/pasted (and future updates can
consolidate more of it).

Also removing some code from the video/voice media channels that's
effectively dead code (vector + registration methods that aren't needed)

Bug: webrtc:12705
Change-Id: I2e14e69fbc489a64fc1e8899aaf1cfc979fe840b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215978
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33847}
This commit is contained in:
Tommi 2021-04-26 20:11:18 +02:00 committed by Commit Bot
parent 553fd3220b
commit a63bee55f2
6 changed files with 47 additions and 107 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
media/engine/webrtc_video_engine.cc
View File

@ -39,7 +39,6 @@
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
@ -704,7 +703,7 @@ WebRtcVideoChannel::WebRtcVideoChannel(
webrtc::VideoDecoderFactory* decoder_factory,
webrtc::VideoBitrateAllocatorFactory* bitrate_allocator_factory)
: VideoMediaChannel(config),
worker_thread_(rtc::Thread::Current()),
worker_thread_(call->worker_thread()),
call_(call),
unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
video_config_(config.video),
@ -2066,7 +2065,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
// TODO(deadbeef): Don't duplicate information between send_params,
// rtp_extensions, options, etc.
const VideoSendParameters& send_params)
: worker_thread_(rtc::Thread::Current()),
: worker_thread_(call->worker_thread()),
ssrcs_(sp.ssrcs),
ssrc_groups_(sp.ssrc_groups),
call_(call),

8
media/engine/webrtc_video_engine.h
View File

@ -43,10 +43,6 @@ class VideoEncoderFactory;
struct MediaConfig;
} // namespace webrtc
namespace rtc {
class Thread;
} // namespace rtc
namespace cricket {
class WebRtcVideoChannel;
@ -402,7 +398,7 @@ class WebRtcVideoChannel : public VideoMediaChannel,
RTC_EXCLUSIVE_LOCKS_REQUIRED(&thread_checker_);
webrtc::SequenceChecker thread_checker_;
rtc::Thread* worker_thread_;
webrtc::TaskQueueBase* const worker_thread_;
const std::vector<uint32_t> ssrcs_ RTC_GUARDED_BY(&thread_checker_);
const std::vector<SsrcGroup> ssrc_groups_ RTC_GUARDED_BY(&thread_checker_);
webrtc::Call* const call_;
@ -553,7 +549,7 @@ class WebRtcVideoChannel : public VideoMediaChannel,
void FillSendAndReceiveCodecStats(VideoMediaInfo* video_media_info)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
rtc::Thread* const worker_thread_;
webrtc::TaskQueueBase* const worker_thread_;
webrtc::ScopedTaskSafety task_safety_;
webrtc::SequenceChecker network_thread_checker_;
webrtc::SequenceChecker thread_checker_;

21
media/engine/webrtc_voice_engine.cc
View File

@ -50,7 +50,6 @@
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/task_utils/to_queued_task.h"
#include "rtc_base/third_party/base64/base64.h"
#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
@ -377,7 +376,7 @@ VoiceMediaChannel* WebRtcVoiceEngine::CreateMediaChannel(
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_RUN_ON(call->worker_thread());
return new WebRtcVoiceMediaChannel(this, config, options, crypto_options,
call);
}
@ -626,19 +625,6 @@ WebRtcVoiceEngine::GetRtpHeaderExtensions() const {
return result;
}
void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK(channel);
channels_.push_back(channel);
}
void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
auto it = absl::c_find(channels_, channel);
RTC_DCHECK(it != channels_.end());
channels_.erase(it);
}
bool WebRtcVoiceEngine::StartAecDump(webrtc::FileWrapper file,
int64_t max_size_bytes) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
@ -1396,18 +1382,16 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(
const webrtc::CryptoOptions& crypto_options,
webrtc::Call* call)
: VoiceMediaChannel(config),
worker_thread_(rtc::Thread::Current()),
worker_thread_(call->worker_thread()),
engine_(engine),
call_(call),
audio_config_(config.audio),
crypto_options_(crypto_options),
audio_red_for_opus_trial_enabled_(
IsEnabled(call->trials(), "WebRTC-Audio-Red-For-Opus")) {
RTC_DCHECK_RUN_ON(worker_thread_);
network_thread_checker_.Detach();
RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
RTC_DCHECK(call);
engine->RegisterChannel(this);
SetOptions(options);
}
@ -1423,7 +1407,6 @@ WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
while (!recv_streams_.empty()) {
RemoveRecvStream(recv_streams_.begin()->first);
}
engine()->UnregisterChannel(this);
}
bool WebRtcVoiceMediaChannel::SetSendParameters(

7
media/engine/webrtc_voice_engine.h
View File

@ -80,12 +80,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
const override;
// For tracking WebRtc channels. Needed because we have to pause them
// all when switching devices.
// May only be called by WebRtcVoiceMediaChannel.
void RegisterChannel(WebRtcVoiceMediaChannel* channel);
void UnregisterChannel(WebRtcVoiceMediaChannel* channel);
// Starts AEC dump using an existing file. A maximum file size in bytes can be
// specified. When the maximum file size is reached, logging is stopped and
// the file is closed. If max_size_bytes is set to <= 0, no limit will be
@ -129,7 +123,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::vector<AudioCodec> send_codecs_;
std::vector<AudioCodec> recv_codecs_;
std::vector<WebRtcVoiceMediaChannel*> channels_;
bool is_dumping_aec_ = false;
bool initialized_ = false;

97
pc/channel.cc
View File

@ -295,7 +295,11 @@ void BaseChannel::Enable(bool enable) {
bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
SetContent_s(content, type);
return InvokeOnWorker<bool>(RTC_FROM_HERE, [this, content, type, error_desc] {
RTC_DCHECK_RUN_ON(worker_thread());
return SetLocalContent_w(content, type, error_desc);
@ -305,13 +309,24 @@ bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
SetContent_s(content, type);
return InvokeOnWorker<bool>(RTC_FROM_HERE, [this, content, type, error_desc] {
RTC_DCHECK_RUN_ON(worker_thread());
return SetRemoteContent_w(content, type, error_desc);
});
}
void BaseChannel::SetContent_s(const MediaContentDescription* content,
SdpType type) {
RTC_DCHECK(content);
if (type == SdpType::kAnswer)
negotiated_header_extensions_ = content->rtp_header_extensions();
}
bool BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) {
TRACE_EVENT0("webrtc", "BaseChannel::SetPayloadTypeDemuxingEnabled");
return InvokeOnWorker<bool>(RTC_FROM_HERE, [this, enabled] {
@ -853,16 +868,8 @@ void BaseChannel::SignalSentPacket_n(const rtc::SentPacket& sent_packet) {
media_channel()->OnPacketSent(sent_packet);
}
void BaseChannel::SetNegotiatedHeaderExtensions_w(
const RtpHeaderExtensions& extensions) {
TRACE_EVENT0("webrtc", __func__);
webrtc::MutexLock lock(&negotiated_header_extensions_lock_);
negotiated_header_extensions_ = extensions;
}
RtpHeaderExtensions BaseChannel::GetNegotiatedRtpHeaderExtensions() const {
RTC_DCHECK_RUN_ON(signaling_thread());
webrtc::MutexLock lock(&negotiated_header_extensions_lock_);
return negotiated_header_extensions_;
}
@ -913,26 +920,19 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local voice description for " << ToString();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in local description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
if (type == SdpType::kAnswer)
SetNegotiatedHeaderExtensions_w(audio->rtp_header_extensions());
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
GetFilteredRtpHeaderExtensions(content->rtp_header_extensions());
// TODO(tommi): There's a hop to the network thread here.
// some of the below is also network thread related.
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
media_channel()->SetExtmapAllowMixed(audio->extmap_allow_mixed());
media_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
AudioRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(
audio, rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(audio->direction()), &recv_params);
content->as_audio(), rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(content->direction()),
&recv_params);
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError(
"Failed to set local audio description recv parameters for m-section "
@ -942,8 +942,8 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
return false;
}
if (webrtc::RtpTransceiverDirectionHasRecv(audio->direction())) {
for (const AudioCodec& codec : audio->codecs()) {
if (webrtc::RtpTransceiverDirectionHasRecv(content->direction())) {
for (const AudioCodec& codec : content->as_audio()->codecs()) {
MaybeAddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
@ -959,7 +959,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
// only give it to the media channel once we have a remote
// description too (without a remote description, we won't be able
// to send them anyway).
if (!UpdateLocalStreams_w(audio->streams(), type, error_desc)) {
if (!UpdateLocalStreams_w(content->as_audio()->streams(), type, error_desc)) {
SafeSetError(
"Failed to set local audio description streams for m-section with "
"mid='" +
@ -980,17 +980,8 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote voice description for " << ToString();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in remote description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
if (type == SdpType::kAnswer)
SetNegotiatedHeaderExtensions_w(audio->rtp_header_extensions());
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
@ -1091,26 +1082,17 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local video description for " << ToString();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find video content in local description.", error_desc);
return false;
}
const VideoContentDescription* video = content->as_video();
if (type == SdpType::kAnswer)
SetNegotiatedHeaderExtensions_w(video->rtp_header_extensions());
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
GetFilteredRtpHeaderExtensions(content->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
media_channel()->SetExtmapAllowMixed(video->extmap_allow_mixed());
media_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
VideoRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(
video, rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(video->direction()), &recv_params);
content->as_video(), rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(content->direction()),
&recv_params);
VideoSendParameters send_params = last_send_params_;
@ -1143,8 +1125,8 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
return false;
}
if (webrtc::RtpTransceiverDirectionHasRecv(video->direction())) {
for (const VideoCodec& codec : video->codecs()) {
if (webrtc::RtpTransceiverDirectionHasRecv(content->direction())) {
for (const VideoCodec& codec : content->as_video()->codecs()) {
MaybeAddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
@ -1170,7 +1152,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
// only give it to the media channel once we have a remote
// description too (without a remote description, we won't be able
// to send them anyway).
if (!UpdateLocalStreams_w(video->streams(), type, error_desc)) {
if (!UpdateLocalStreams_w(content->as_video()->streams(), type, error_desc)) {
SafeSetError(
"Failed to set local video description streams for m-section with "
"mid='" +
@ -1191,17 +1173,8 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote video description for " << ToString();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find video content in remote description.", error_desc);
return false;
}
const VideoContentDescription* video = content->as_video();
if (type == SdpType::kAnswer)
SetNegotiatedHeaderExtensions_w(video->rtp_header_extensions());
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());

16
pc/channel.h
View File

@ -59,7 +59,6 @@
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/socket.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
@ -306,9 +305,6 @@ class BaseChannel : public ChannelInterface,
// Return description of media channel to facilitate logging
std::string ToString() const;
void SetNegotiatedHeaderExtensions_w(const RtpHeaderExtensions& extensions)
RTC_RUN_ON(worker_thread());
// ChannelInterface overrides
RtpHeaderExtensions GetNegotiatedRtpHeaderExtensions() const override;
@ -316,6 +312,8 @@ class BaseChannel : public ChannelInterface,
bool ConnectToRtpTransport() RTC_RUN_ON(network_thread());
void DisconnectFromRtpTransport() RTC_RUN_ON(network_thread());
void SignalSentPacket_n(const rtc::SentPacket& sent_packet);
void SetContent_s(const MediaContentDescription* content,
webrtc::SdpType type) RTC_RUN_ON(signaling_thread());
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
@ -382,13 +380,11 @@ class BaseChannel : public ChannelInterface,
// This object is not owned by the channel so it must outlive it.
rtc::UniqueRandomIdGenerator* const ssrc_generator_;
// |negotiated_header_extensions_| is read on the signaling thread, but
// written on the worker thread while being sync-invoked from the signal
// thread in SdpOfferAnswerHandler::PushdownMediaDescription(). Hence the lock
// isn't strictly needed, but it's anyway placed here for future safeness.
mutable webrtc::Mutex negotiated_header_extensions_lock_;
// |negotiated_header_extensions_| is read and written to on the signaling
// thread from the SdpOfferAnswerHandler class (e.g.
// PushdownMediaDescription().
RtpHeaderExtensions negotiated_header_extensions_
RTC_GUARDED_BY(negotiated_header_extensions_lock_);
RTC_GUARDED_BY(signaling_thread());
};
// VoiceChannel is a specialization that adds support for early media, DTMF,