admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Remove more dead code from BaseChannel

Browse Source 
This removes the following methods:
- SetAudioSend (directly accessed through MediaChannel now)
- "Early Media" (feature not used)
- GetStats (directly accessed through MediaChannel now)

Bug: None
Change-Id: Ifd075d030b0f5f41e94918979891592a731d5a91
Reviewed-on: https://webrtc-review.googlesource.com/59500
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22298}
This commit is contained in:
Steve Anton 2018-03-05 11:23:09 -08:00 committed by Commit Bot
parent 0867260590
commit 0807d152d5
3 changed files with 4 additions and 142 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

68
pc/channel.cc
View File

@ -47,8 +47,7 @@ struct SendPacketMessageData : public rtc::MessageData {
} // namespace
enum {
MSG_EARLYMEDIATIMEOUT = 1,
MSG_SEND_RTP_PACKET,
MSG_SEND_RTP_PACKET = 1,
MSG_SEND_RTCP_PACKET,
MSG_READYTOSENDDATA,
MSG_DATARECEIVED,
@ -1155,47 +1154,6 @@ VoiceChannel::~VoiceChannel() {
Deinit();
}
bool VoiceChannel::SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE, Bind(&VoiceMediaChannel::SetAudioSend, media_channel(),
ssrc, enable, options, source));
}
// TODO(juberti): Handle early media the right way. We should get an explicit
// ringing message telling us to start playing local ringback, which we cancel
// if any early media actually arrives. For now, we do the opposite, which is
// to wait 1 second for early media, and start playing local ringback if none
// arrives.
void VoiceChannel::SetEarlyMedia(bool enable) {
if (enable) {
// Start the early media timeout
worker_thread()->PostDelayed(RTC_FROM_HERE, kEarlyMediaTimeout, this,
MSG_EARLYMEDIATIMEOUT);
} else {
// Stop the timeout if currently going.
worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT);
}
}
bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats,
media_channel(), stats));
}
void VoiceChannel::OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
BaseChannel::OnPacketReceived(rtcp, packet, packet_time);
// Set a flag when we've received an RTP packet. If we're waiting for early
// media, this will disable the timeout.
if (!received_media_ && !rtcp) {
received_media_ = true;
}
}
void BaseChannel::UpdateMediaSendRecvState() {
RTC_DCHECK(network_thread_->IsCurrent());
invoker_.AsyncInvoke<void>(
@ -1319,25 +1277,6 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
return true;
}
void VoiceChannel::HandleEarlyMediaTimeout() {
// This occurs on the main thread, not the worker thread.
if (!received_media_) {
RTC_LOG(LS_INFO) << "No early media received before timeout";
SignalEarlyMediaTimeout(this);
}
}
void VoiceChannel::OnMessage(rtc::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_EARLYMEDIATIMEOUT:
HandleEarlyMediaTimeout();
break;
default:
BaseChannel::OnMessage(pmsg);
break;
}
}
VideoChannel::VideoChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
@ -1378,11 +1317,6 @@ void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
media_channel(), bwe_info));
}
bool VideoChannel::GetStats(VideoMediaInfo* stats) {
return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VideoMediaChannel::GetStats,
media_channel(), stats));
}
bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {

34
pc/channel.h
View File

@ -265,9 +265,9 @@ class BaseChannel
void HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
// TODO(zstein): packet can be const once the RtpTransport handles protection.
virtual void OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
void OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
void ProcessPacket(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
const rtc::PacketTime& packet_time);
@ -450,27 +450,11 @@ class VoiceChannel : public BaseChannel {
bool srtp_required);
~VoiceChannel();
// Configure sending media on the stream with SSRC |ssrc|
// If there is only one sending stream SSRC 0 can be used.
bool SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source);
// downcasts a MediaChannel
VoiceMediaChannel* media_channel() const override {
return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
}
void SetEarlyMedia(bool enable);
// This signal is emitted when we have gone a period of time without
// receiving early media. When received, a UI should start playing its
// own ringing sound
sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;
// Get statistics about the current media session.
bool GetStats(VoiceMediaInfo* stats);
webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
webrtc::RTCError SetRtpSendParameters_w(uint32_t ssrc,
webrtc::RtpParameters parameters);
@ -478,9 +462,6 @@ class VoiceChannel : public BaseChannel {
private:
// overrides from BaseChannel
void OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) override;
void UpdateMediaSendRecvState_w() override;
bool SetLocalContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
@ -488,12 +469,6 @@ class VoiceChannel : public BaseChannel {
bool SetRemoteContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
void HandleEarlyMediaTimeout();
void OnMessage(rtc::Message* pmsg) override;
static const int kEarlyMediaTimeout = 1000;
bool received_media_ = false;
// Last AudioSendParameters sent down to the media_channel() via
// SetSendParameters.
@ -521,8 +496,6 @@ class VideoChannel : public BaseChannel {
}
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
// Get statistics about the current media session.
bool GetStats(VideoMediaInfo* stats);
cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_VIDEO; }
@ -535,7 +508,6 @@ class VideoChannel : public BaseChannel {
bool SetRemoteContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
bool GetStats_w(VideoMediaInfo* stats);
// Last VideoSendParameters sent down to the media_channel() via
// SetSendParameters.

44
pc/channel_unittest.cc
View File

@ -2166,28 +2166,6 @@ TEST_F(VoiceChannelSingleThreadTest, TestPlayoutAndSendingStates) {
Base::TestPlayoutAndSendingStates();
}
TEST_F(VoiceChannelSingleThreadTest, TestMuteStream) {
CreateChannels(0, 0);
// Test that we can Mute the default channel even though the sending SSRC
// is unknown.
EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
EXPECT_TRUE(channel1_->SetAudioSend(0, false, nullptr, nullptr));
EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
EXPECT_TRUE(channel1_->SetAudioSend(0, true, nullptr, nullptr));
EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
// Test that we can not mute an unknown SSRC.
EXPECT_FALSE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
SendInitiate();
// After the local session description has been set, we can mute a stream
// with its SSRC.
EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
EXPECT_TRUE(media_channel1_->IsStreamMuted(kSsrc1));
EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
}
TEST_F(VoiceChannelSingleThreadTest, TestMediaContentDirection) {
Base::TestMediaContentDirection();
}
@ -2427,28 +2405,6 @@ TEST_F(VoiceChannelDoubleThreadTest, TestPlayoutAndSendingStates) {
Base::TestPlayoutAndSendingStates();
}
TEST_F(VoiceChannelDoubleThreadTest, TestMuteStream) {
CreateChannels(0, 0);
// Test that we can Mute the default channel even though the sending SSRC
// is unknown.
EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
EXPECT_TRUE(channel1_->SetAudioSend(0, false, nullptr, nullptr));
EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
EXPECT_TRUE(channel1_->SetAudioSend(0, true, nullptr, nullptr));
EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
// Test that we can not mute an unknown SSRC.
EXPECT_FALSE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
SendInitiate();
// After the local session description has been set, we can mute a stream
// with its SSRC.
EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
EXPECT_TRUE(media_channel1_->IsStreamMuted(kSsrc1));
EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
}
TEST_F(VoiceChannelDoubleThreadTest, TestMediaContentDirection) {
Base::TestMediaContentDirection();
}