Delete ChannelSend::RegisterTransport, replacing by construction argument
Bug: webrtc:9719
Change-Id: If3960de660cfa7a65c8bf9375ceb0af0a67d376c
Reviewed-on: https://webrtc-review.googlesource.com/c/111256
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25784}diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 082c557..75e6efb 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -111,6 +111,7 @@
voe::CreateChannelSend(worker_queue,
module_process_thread,
config.media_transport,
+ config.send_transport,
rtcp_rtt_stats,
event_log,
config.frame_encryptor,
@@ -171,7 +172,6 @@
RTC_DCHECK(!sending_);
if (rtp_transport_) {
rtp_transport_->DeRegisterPacketFeedbackObserver(this);
- channel_send_->RegisterTransport(nullptr);
channel_send_->ResetSenderCongestionControlObjects();
}
}
@@ -214,6 +214,10 @@
const auto& channel_send = stream->channel_send_;
const auto& old_config = stream->config_;
+ // Configuration parameters which cannot be changed.
+ RTC_DCHECK(first_time ||
+ old_config.send_transport == new_config.send_transport);
+
if (first_time || old_config.rtp.ssrc != new_config.rtp.ssrc) {
channel_send->SetLocalSSRC(new_config.rtp.ssrc);
if (stream->suspended_rtp_state_) {
@@ -224,10 +228,6 @@
channel_send->SetRTCP_CNAME(new_config.rtp.c_name);
}
- if (first_time || new_config.send_transport != old_config.send_transport) {
- channel_send->RegisterTransport(new_config.send_transport);
- }
-
// Enable the frame encryptor if a new frame encryptor has been provided.
if (first_time || new_config.frame_encryptor != old_config.frame_encryptor) {
channel_send->SetFrameEncryptor(new_config.frame_encryptor);
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 13e5884..e400ada 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -211,7 +211,6 @@
.Times(1);
}
EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1);
- EXPECT_CALL(*channel_send_, RegisterTransport(nullptr)).Times(2);
}
void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 31fe25f..c458fe4 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -91,6 +91,7 @@
ChannelSend(rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
+ Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
@@ -115,7 +116,6 @@
int GetBitrate() const override;
// Network
- void RegisterTransport(Transport* transport) override;
bool ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
// Muting, Volume and Level.
@@ -254,7 +254,7 @@
// uses
ProcessThread* const _moduleProcessThreadPtr;
- Transport* _transportPtr; // WebRtc socket or external transport
+ 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_);
@@ -683,6 +683,7 @@
ChannelSend::ChannelSend(rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
+ Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
@@ -694,7 +695,7 @@
// random offset
send_sequence_number_(0),
_moduleProcessThreadPtr(module_process_thread),
- _transportPtr(NULL),
+ _transportPtr(rtp_transport),
input_mute_(false),
previous_frame_muted_(false),
_includeAudioLevelIndication(false),
@@ -953,12 +954,6 @@
});
}
-void ChannelSend::RegisterTransport(Transport* transport) {
- RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- rtc::CritScope cs(&_callbackCritSect);
- _transportPtr = transport;
-}
-
// TODO(nisse): Delete always-true return value.
bool ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
// May be called on either worker thread or network thread.
@@ -1344,6 +1339,7 @@
rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
+ Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
@@ -1351,9 +1347,9 @@
bool extmap_allow_mixed,
int rtcp_report_interval_ms) {
return absl::make_unique<ChannelSend>(
- encoder_queue, module_process_thread, media_transport, rtcp_rtt_stats,
- rtc_event_log, frame_encryptor, crypto_options, extmap_allow_mixed,
- rtcp_report_interval_ms);
+ encoder_queue, module_process_thread, media_transport, rtp_transport,
+ rtcp_rtt_stats, rtc_event_log, frame_encryptor, crypto_options,
+ extmap_allow_mixed, rtcp_report_interval_ms);
}
} // namespace voe
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 231fda6..083e9a6 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -55,7 +55,6 @@
public:
virtual ~ChannelSendInterface() = default;
- virtual void RegisterTransport(Transport* transport) = 0;
virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0;
virtual CallSendStatistics GetRTCPStatistics() const = 0;
@@ -115,6 +114,7 @@
rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
+ Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index fe49359..eee25c5 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -90,7 +90,6 @@
MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
MOCK_METHOD1(OnBitrateAllocation, void(BitrateAllocationUpdate update));
MOCK_METHOD1(SetInputMute, void(bool muted));
- MOCK_METHOD1(RegisterTransport, void(Transport* transport));
MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
// GMock doesn't like move-only types, like std::unique_ptr.
virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) {
