Remove Transport implementation from ChannelSend
Avoids taking a lock for each outgoing packet.
Bug: none
Change-Id: I54defbf07097ea8032b556b6900ca58c7486c3d9
Reviewed-on: https://webrtc-review.googlesource.com/c/112123
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26038}diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 7c7db79..fa21141 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -78,7 +78,6 @@
class ChannelSend
: public ChannelSendInterface,
- public Transport,
public OverheadObserver,
public AudioPacketizationCallback, // receive encoded packets from the
// ACM
@@ -186,12 +185,6 @@
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) override;
- // From Transport (called by the RTP/RTCP module)
- bool SendRtp(const uint8_t* data,
- size_t len,
- const PacketOptions& packet_options) override;
- bool SendRtcp(const uint8_t* data, size_t len) override;
-
// From OverheadObserver in the RTP/RTCP module
void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
@@ -238,7 +231,6 @@
// audio thread to another, but access is still sequential.
rtc::RaceChecker audio_thread_race_checker_;
- rtc::CriticalSection _callbackCritSect;
rtc::CriticalSection volume_settings_critsect_;
bool sending_ RTC_GUARDED_BY(&worker_thread_checker_) = false;
@@ -254,7 +246,6 @@
// uses
ProcessThread* const _moduleProcessThreadPtr;
- Transport* const _transportPtr; // WebRtc socket or external transport
RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_);
@@ -641,45 +632,6 @@
return 0;
}
-bool ChannelSend::SendRtp(const uint8_t* data,
- size_t len,
- const PacketOptions& options) {
- // We should not be sending RTP packets if media transport is available.
- RTC_CHECK(!media_transport());
-
- rtc::CritScope cs(&_callbackCritSect);
-
- if (_transportPtr == NULL) {
- RTC_DLOG(LS_ERROR)
- << "ChannelSend::SendPacket() failed to send RTP packet due to"
- << " invalid transport object";
- return false;
- }
-
- if (!_transportPtr->SendRtp(data, len, options)) {
- RTC_DLOG(LS_ERROR) << "ChannelSend::SendPacket() RTP transmission failed";
- return false;
- }
- return true;
-}
-
-bool ChannelSend::SendRtcp(const uint8_t* data, size_t len) {
- rtc::CritScope cs(&_callbackCritSect);
- if (_transportPtr == NULL) {
- RTC_DLOG(LS_ERROR)
- << "ChannelSend::SendRtcp() failed to send RTCP packet due to"
- << " invalid transport object";
- return false;
- }
-
- int n = _transportPtr->SendRtcp(data, len);
- if (n < 0) {
- RTC_DLOG(LS_ERROR) << "ChannelSend::SendRtcp() transmission failed";
- return false;
- }
- return true;
-}
-
ChannelSend::ChannelSend(rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
@@ -695,7 +647,6 @@
// random offset
send_sequence_number_(0),
_moduleProcessThreadPtr(module_process_thread),
- _transportPtr(rtp_transport),
input_mute_(false),
previous_frame_muted_(false),
_includeAudioLevelIndication(false),
@@ -732,7 +683,7 @@
}
configuration.audio = true;
- configuration.outgoing_transport = this;
+ configuration.outgoing_transport = rtp_transport;
configuration.paced_sender = rtp_packet_sender_proxy_.get();
configuration.transport_sequence_number_allocator =
diff --git a/call/call_unittest.cc b/call/call_unittest.cc
index 770f255..1739036 100644
--- a/call/call_unittest.cc
+++ b/call/call_unittest.cc
@@ -63,7 +63,8 @@
TEST(CallTest, CreateDestroy_AudioSendStream) {
CallHelper call;
- AudioSendStream::Config config(/*send_transport=*/nullptr,
+ MockTransport send_transport;
+ AudioSendStream::Config config(&send_transport,
/*media_transport=*/nullptr);
config.rtp.ssrc = 42;
AudioSendStream* stream = call->CreateAudioSendStream(config);
@@ -86,7 +87,8 @@
TEST(CallTest, CreateDestroy_AudioSendStreams) {
CallHelper call;
- AudioSendStream::Config config(/*send_transport=*/nullptr,
+ MockTransport send_transport;
+ AudioSendStream::Config config(&send_transport,
/*media_transport=*/nullptr);
std::list<AudioSendStream*> streams;
for (int i = 0; i < 2; ++i) {
@@ -145,7 +147,8 @@
AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
EXPECT_NE(recv_stream, nullptr);
- AudioSendStream::Config send_config(/*send_transport=*/nullptr,
+ MockTransport send_transport;
+ AudioSendStream::Config send_config(&send_transport,
/*media_transport=*/nullptr);
send_config.rtp.ssrc = 777;
AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
@@ -164,7 +167,8 @@
TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
CallHelper call;
- AudioSendStream::Config send_config(/*send_transport=*/nullptr,
+ MockTransport send_transport;
+ AudioSendStream::Config send_config(&send_transport,
/*media_transport=*/nullptr);
send_config.rtp.ssrc = 777;
AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
@@ -268,7 +272,8 @@
CallHelper call;
auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
- AudioSendStream::Config config(/*send_transport=*/nullptr,
+ MockTransport send_transport;
+ AudioSendStream::Config config(&send_transport,
/*media_transport=*/nullptr);
config.rtp.ssrc = ssrc;
AudioSendStream* stream = call->CreateAudioSendStream(config);
@@ -296,7 +301,10 @@
webrtc::FakeMediaTransport fake_media_transport(settings);
EXPECT_EQ(0, fake_media_transport.target_rate_observers_size());
- AudioSendStream::Config config(/*send_transport=*/nullptr,
+ // TODO(solenberg): This test shouldn't require a Transport, but currently
+ // RTCPSender requires one.
+ MockTransport send_transport;
+ AudioSendStream::Config config(&send_transport,
/*media_transport=*/&fake_media_transport);
call->MediaTransportChange(&fake_media_transport);
