Tweak ChannelSend interface, to make it closer to ChannelSendProxy
This is a preparation for deleting ChannelSendProxy. Signature is
changed on a couple of methods. Unused methods
EnableAudioNetworkAdaptor, DisableAudioNetworkAdaptor,
SetReceiverFrameLengthRange and RtpRtcpModulePtr are deleted. Some
methods are demoted to private: SendData, SendRtp, SendRtcp,
PreferredSampleRate, Sending, and OnOverheadChanged.
Bug: webrtc:9801
Change-Id: I982e72418a32e66fb5de410350b1bfebd9a3219c
Reviewed-on: https://webrtc-review.googlesource.com/c/110605
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25666}diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index a4a4977..09482a4 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -33,6 +33,7 @@
#include "rtc_base/format_macros.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/thread_checker.h"
@@ -557,10 +558,8 @@
// End of modules shutdown
}
-int32_t ChannelSend::StartSend() {
- if (channel_state_.Get().sending) {
- return 0;
- }
+void ChannelSend::StartSend() {
+ RTC_DCHECK(!channel_state_.Get().sending);
channel_state_.SetSending(true);
// Resume the previous sequence number which was reset by StopSend(). This
@@ -569,19 +568,13 @@
_rtpRtcpModule->SetSequenceNumber(send_sequence_number_);
}
_rtpRtcpModule->SetSendingMediaStatus(true);
- if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
- RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending";
- _rtpRtcpModule->SetSendingMediaStatus(false);
- rtc::CritScope cs(&_callbackCritSect);
- channel_state_.SetSending(false);
- return -1;
- }
+ int ret = _rtpRtcpModule->SetSendingStatus(true);
+ RTC_DCHECK_EQ(0, ret);
{
// It is now OK to start posting tasks to the encoder task queue.
rtc::CritScope cs(&encoder_queue_lock_);
encoder_queue_is_active_ = true;
}
- return 0;
}
void ChannelSend::StopSend() {
@@ -716,47 +709,20 @@
});
}
-bool ChannelSend::EnableAudioNetworkAdaptor(const std::string& config_string) {
- bool success = false;
- audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
- if (*encoder) {
- success =
- (*encoder)->EnableAudioNetworkAdaptor(config_string, event_log_);
- }
- });
- return success;
-}
-
-void ChannelSend::DisableAudioNetworkAdaptor() {
- audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
- if (*encoder)
- (*encoder)->DisableAudioNetworkAdaptor();
- });
-}
-
-void ChannelSend::SetReceiverFrameLengthRange(int min_frame_length_ms,
- int max_frame_length_ms) {
- audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
- if (*encoder) {
- (*encoder)->SetReceiverFrameLengthRange(min_frame_length_ms,
- max_frame_length_ms);
- }
- });
-}
-
void ChannelSend::RegisterTransport(Transport* transport) {
rtc::CritScope cs(&_callbackCritSect);
_transportPtr = transport;
}
-int32_t ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
+// TODO(nisse): Delete always-true return value.
+bool ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
// Deliver RTCP packet to RTP/RTCP module for parsing
_rtpRtcpModule->IncomingRtcpPacket(data, length);
int64_t rtt = GetRTT();
if (rtt == 0) {
// Waiting for valid RTT.
- return 0;
+ return true;
}
int64_t nack_window_ms = rtt;
@@ -772,8 +738,7 @@
if (*encoder)
(*encoder)->OnReceivedRtt(rtt);
});
-
- return 0;
+ return true;
}
void ChannelSend::SetInputMute(bool enable) {
@@ -786,24 +751,24 @@
return input_mute_;
}
-int ChannelSend::SendTelephoneEventOutband(int event, int duration_ms) {
+bool ChannelSend::SendTelephoneEventOutband(int event, int duration_ms) {
RTC_DCHECK_LE(0, event);
RTC_DCHECK_GE(255, event);
RTC_DCHECK_LE(0, duration_ms);
RTC_DCHECK_GE(65535, duration_ms);
if (!Sending()) {
- return -1;
+ return false;
}
if (_rtpRtcpModule->SendTelephoneEventOutband(
event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
RTC_DLOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event";
- return -1;
+ return false;
}
- return 0;
+ return true;
}
-int ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
- int payload_frequency) {
+bool ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
+ int payload_frequency) {
RTC_DCHECK_LE(0, payload_type);
RTC_DCHECK_GE(127, payload_type);
CodecInst codec = {0};
@@ -816,23 +781,20 @@
RTC_DLOG(LS_ERROR)
<< "SetSendTelephoneEventPayloadType() failed to register "
"send payload type";
- return -1;
+ return false;
}
}
- return 0;
+ return true;
}
-int ChannelSend::SetLocalSSRC(unsigned int ssrc) {
- if (channel_state_.Get().sending) {
- RTC_DLOG(LS_ERROR) << "SetLocalSSRC() already sending";
- return -1;
- }
+void ChannelSend::SetLocalSSRC(unsigned int ssrc) {
+ RTC_DCHECK(!channel_state_.Get().sending);
+
if (media_transport_) {
rtc::CritScope cs(&media_transport_lock_);
media_transport_channel_id_ = ssrc;
}
_rtpRtcpModule->SetSSRC(ssrc);
- return 0;
}
void ChannelSend::SetMid(const std::string& mid, int extension_id) {
@@ -845,10 +807,10 @@
_rtpRtcpModule->SetExtmapAllowMixed(extmap_allow_mixed);
}
-int ChannelSend::SetSendAudioLevelIndicationStatus(bool enable,
- unsigned char id) {
+void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
_includeAudioLevelIndication = enable;
- return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
+ int ret = SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
+ RTC_DCHECK_EQ(0, ret);
}
void ChannelSend::EnableSendTransportSequenceNumber(int id) {
@@ -895,31 +857,23 @@
_rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
}
-int ChannelSend::SetRTCP_CNAME(const char cName[256]) {
- if (_rtpRtcpModule->SetCNAME(cName) != 0) {
- RTC_DLOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME";
- return -1;
- }
- return 0;
+void ChannelSend::SetRTCP_CNAME(absl::string_view c_name) {
+ // Note: SetCNAME() accepts a c string of length at most 255.
+ const std::string c_name_limited(c_name.substr(0, 255));
+ int ret = _rtpRtcpModule->SetCNAME(c_name_limited.c_str()) != 0;
+ RTC_DCHECK_EQ(0, ret) << "SetRTCP_CNAME() failed to set RTCP CNAME";
}
-int ChannelSend::GetRemoteRTCPReportBlocks(
- std::vector<ReportBlock>* report_blocks) {
- if (report_blocks == NULL) {
- RTC_DLOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks.";
- return -1;
- }
-
+std::vector<ReportBlock> ChannelSend::GetRemoteRTCPReportBlocks() const {
// Get the report blocks from the latest received RTCP Sender or Receiver
// Report. Each element in the vector contains the sender's SSRC and a
// report block according to RFC 3550.
std::vector<RTCPReportBlock> rtcp_report_blocks;
- if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks) != 0) {
- return -1;
- }
- if (rtcp_report_blocks.empty())
- return 0;
+ int ret = _rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks);
+ RTC_DCHECK_EQ(0, ret);
+
+ std::vector<ReportBlock> report_blocks;
std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin();
for (; it != rtcp_report_blocks.end(); ++it) {
@@ -933,19 +887,15 @@
report_block.interarrival_jitter = it->jitter;
report_block.last_SR_timestamp = it->last_sender_report_timestamp;
report_block.delay_since_last_SR = it->delay_since_last_sender_report;
- report_blocks->push_back(report_block);
+ report_blocks.push_back(report_block);
}
- return 0;
+ return report_blocks;
}
-int ChannelSend::GetRTPStatistics(CallSendStatistics& stats) {
- // --- RtcpStatistics
-
- // --- RTT
+CallSendStatistics ChannelSend::GetRTCPStatistics() const {
+ CallSendStatistics stats = {0};
stats.rttMs = GetRTT();
- // --- Data counters
-
size_t bytesSent(0);
uint32_t packetsSent(0);
@@ -958,7 +908,7 @@
stats.bytesSent = bytesSent;
stats.packetsSent = packetsSent;
- return 0;
+ return stats;
}
void ChannelSend::SetNACKStatus(bool enable, int maxNumberOfPackets) {
@@ -1063,11 +1013,14 @@
int ChannelSend::SetSendRtpHeaderExtension(bool enable,
RTPExtensionType type,
- unsigned char id) {
+ int id) {
int error = 0;
_rtpRtcpModule->DeregisterSendRtpHeaderExtension(type);
if (enable) {
- error = _rtpRtcpModule->RegisterSendRtpHeaderExtension(type, id);
+ // TODO(nisse): RtpRtcp::RegisterSendRtpHeaderExtension to take an int
+ // argument. Currently it wants an uint8_t.
+ error = _rtpRtcpModule->RegisterSendRtpHeaderExtension(
+ type, rtc::dchecked_cast<uint8_t>(id));
}
return error;
}