Remove some dead code in ViEChannel.
BUG=1695
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54429004
Cr-Commit-Position: refs/heads/master@{#9203}
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index f0cc56c..86fe598 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -130,8 +130,7 @@
nack_history_size_sender_(kSendSidePacketHistorySize),
max_nack_reordering_threshold_(kMaxPacketAgeToNack),
pre_render_callback_(NULL),
- report_block_stats_sender_(new ReportBlockStats()),
- report_block_stats_receiver_(new ReportBlockStats()) {
+ report_block_stats_sender_(new ReportBlockStats()) {
RtpRtcp::Configuration configuration = CreateRtpRtcpConfiguration();
configuration.remote_bitrate_estimator = remote_bitrate_estimator;
configuration.receive_statistics = vie_receiver_.GetReceiveStatistics();
@@ -264,11 +263,6 @@
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
rtcp_counter.UniqueNackRequestsInPercent());
}
- int fraction_lost = report_block_stats_receiver_->FractionLostInPercent();
- if (fraction_lost != -1) {
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
- fraction_lost);
- }
}
StreamDataCounters rtp;
@@ -574,13 +568,6 @@
return 0;
}
-int32_t ViEChannel::GetReceiveCodec(VideoCodec* video_codec) {
- if (vcm_->ReceiveCodec(video_codec) != 0) {
- return -1;
- }
- return 0;
-}
-
int32_t ViEChannel::RegisterCodecObserver(ViEDecoderObserver* observer) {
CriticalSectionScoped cs(callback_cs_.get());
if (observer) {
@@ -675,10 +662,6 @@
rtp_rtcp_->SetRTCPStatus(rtcp_mode);
}
-RTCPMethod ViEChannel::GetRTCPMode() const {
- return rtp_rtcp_->RTCP();
-}
-
int32_t ViEChannel::SetNACKStatus(const bool enable) {
// Update the decoding VCM.
if (vcm_->SetVideoProtection(kProtectionNack, enable) != VCM_OK) {
@@ -1022,17 +1005,6 @@
return 0;
}
-int32_t ViEChannel::GetRemoteCSRC(uint32_t CSRCs[kRtpCsrcSize]) {
- uint32_t arrayCSRC[kRtpCsrcSize];
- memset(arrayCSRC, 0, sizeof(arrayCSRC));
-
- int num_csrcs = vie_receiver_.GetCsrcs(arrayCSRC);
- if (num_csrcs > 0) {
- memcpy(CSRCs, arrayCSRC, num_csrcs * sizeof(uint32_t));
- }
- return 0;
-}
-
int ViEChannel::SetRtxSendPayloadType(int payload_type,
int associated_payload_type) {
rtp_rtcp_->SetRtxSendPayloadType(payload_type, associated_payload_type);
@@ -1209,38 +1181,6 @@
}
}
-// TODO(holmer): This is a bad function name as it implies that it returns the
-// received RTCP, while it actually returns the statistics which will be sent
-// in the RTCP.
-int32_t ViEChannel::GetReceivedRtcpStatistics(uint16_t* fraction_lost,
- uint32_t* cumulative_lost,
- uint32_t* extended_max,
- uint32_t* jitter_samples,
- int64_t* rtt_ms) {
- uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
- StreamStatistician* statistician =
- vie_receiver_.GetReceiveStatistics()->GetStatistician(remote_ssrc);
- RtcpStatistics receive_stats;
- if (!statistician || !statistician->GetStatistics(
- &receive_stats, rtp_rtcp_->RTCP() == kRtcpOff)) {
- return -1;
- }
- *fraction_lost = receive_stats.fraction_lost;
- *cumulative_lost = receive_stats.cumulative_lost;
- *extended_max = receive_stats.extended_max_sequence_number;
- *jitter_samples = receive_stats.jitter;
-
- // TODO(asapersson): Change report_block_stats to not rely on
- // GetReceivedRtcpStatistics to be called.
- report_block_stats_receiver_->Store(receive_stats, remote_ssrc, 0);
-
- int64_t dummy = 0;
- int64_t rtt = 0;
- rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy);
- *rtt_ms = rtt;
- return 0;
-}
-
void ViEChannel::RegisterReceiveChannelRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) {
vie_receiver_.GetReceiveStatistics()->RegisterRtcpStatisticsCallback(
@@ -1253,42 +1193,6 @@
rtcp_packet_type_counter_observer_.Set(observer);
}
-int32_t ViEChannel::GetRtpStatistics(size_t* bytes_sent,
- uint32_t* packets_sent,
- size_t* bytes_received,
- uint32_t* packets_received) const {
- StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()->
- GetStatistician(vie_receiver_.GetRemoteSsrc());
- *bytes_received = 0;
- *packets_received = 0;
- if (statistician)
- statistician->GetDataCounters(bytes_received, packets_received);
- if (rtp_rtcp_->DataCountersRTP(bytes_sent, packets_sent) != 0) {
- return -1;
- }
- CriticalSectionScoped cs(rtp_rtcp_cs_.get());
- for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- size_t bytes_sent_temp = 0;
- uint32_t packets_sent_temp = 0;
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->DataCountersRTP(&bytes_sent_temp, &packets_sent_temp);
- *bytes_sent += bytes_sent_temp;
- *packets_sent += packets_sent_temp;
- }
- for (std::list<RtpRtcp*>::const_iterator it = removed_rtp_rtcp_.begin();
- it != removed_rtp_rtcp_.end(); ++it) {
- size_t bytes_sent_temp = 0;
- uint32_t packets_sent_temp = 0;
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->DataCountersRTP(&bytes_sent_temp, &packets_sent_temp);
- *bytes_sent += bytes_sent_temp;
- *packets_sent += packets_sent_temp;
- }
- return 0;
-}
-
void ViEChannel::GetSendStreamDataCounters(
StreamDataCounters* rtp_counters,
StreamDataCounters* rtx_counters) const {
@@ -1380,59 +1284,6 @@
*packet_counter = counter;
}
-void ViEChannel::GetBandwidthUsage(uint32_t* total_bitrate_sent,
- uint32_t* video_bitrate_sent,
- uint32_t* fec_bitrate_sent,
- uint32_t* nackBitrateSent) const {
- rtp_rtcp_->BitrateSent(total_bitrate_sent, video_bitrate_sent,
- fec_bitrate_sent, nackBitrateSent);
- CriticalSectionScoped cs(rtp_rtcp_cs_.get());
- for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end(); it++) {
- uint32_t stream_rate = 0;
- uint32_t video_rate = 0;
- uint32_t fec_rate = 0;
- uint32_t nackRate = 0;
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->BitrateSent(&stream_rate, &video_rate, &fec_rate, &nackRate);
- *total_bitrate_sent += stream_rate;
- *video_bitrate_sent += video_rate;
- *fec_bitrate_sent += fec_rate;
- *nackBitrateSent += nackRate;
- }
-}
-
-bool ViEChannel::GetSendSideDelay(int* avg_send_delay,
- int* max_send_delay) const {
- *avg_send_delay = 0;
- *max_send_delay = 0;
- bool valid_estimate = false;
- int num_send_delays = 0;
- if (rtp_rtcp_->GetSendSideDelay(avg_send_delay, max_send_delay)) {
- ++num_send_delays;
- valid_estimate = true;
- }
- CriticalSectionScoped cs(rtp_rtcp_cs_.get());
- for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end(); it++) {
- RtpRtcp* rtp_rtcp = *it;
- int sub_stream_avg_delay = 0;
- int sub_stream_max_delay = 0;
- if (rtp_rtcp->GetSendSideDelay(&sub_stream_avg_delay,
- &sub_stream_max_delay)) {
- *avg_send_delay += sub_stream_avg_delay;
- *max_send_delay = std::max(*max_send_delay, sub_stream_max_delay);
- ++num_send_delays;
- }
- }
- if (num_send_delays > 0) {
- valid_estimate = true;
- *avg_send_delay = *avg_send_delay / num_send_delays;
- *avg_send_delay = (*avg_send_delay + num_send_delays / 2) / num_send_delays;
- }
- return valid_estimate;
-}
-
void ViEChannel::RegisterSendSideDelayObserver(
SendSideDelayObserver* observer) {
send_side_delay_observer_.Set(observer);
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index e5aed5c..05550ca 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -125,7 +125,6 @@
// type has changed and we should start a new RTP stream.
int32_t SetSendCodec(const VideoCodec& video_codec, bool new_stream = true);
int32_t SetReceiveCodec(const VideoCodec& video_codec);
- int32_t GetReceiveCodec(VideoCodec* video_codec);
int32_t RegisterCodecObserver(ViEDecoderObserver* observer);
// Registers an external decoder. |buffered_rendering| means that the decoder
// will render frames after decoding according to the render timestamp
@@ -152,7 +151,6 @@
int32_t SetSignalPacketLossStatus(bool enable, bool only_key_frames);
void SetRTCPMode(const RTCPMethod rtcp_mode);
- RTCPMethod GetRTCPMode() const;
int32_t SetNACKStatus(const bool enable);
int32_t SetFECStatus(const bool enable,
const unsigned char payload_typeRED,
@@ -169,7 +167,6 @@
int SetReceiveTimestampOffsetStatus(bool enable, int id);
int SetSendAbsoluteSendTimeStatus(bool enable, int id);
int SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
- bool GetReceiveAbsoluteSendTimeStatus() const;
int SetSendVideoRotationStatus(bool enable, int id);
int SetReceiveVideoRotationStatus(bool enable, int id);
void SetRtcpXrRrtrStatus(bool enable);
@@ -188,9 +185,6 @@
// Gets SSRC for the incoming stream.
int32_t GetRemoteSSRC(uint32_t* ssrc);
- // Gets the CSRC for the incoming stream.
- int32_t GetRemoteCSRC(uint32_t CSRCs[kRtpCsrcSize]);
-
int SetRtxSendPayloadType(int payload_type, int associated_payload_type);
void SetRtxReceivePayloadType(int payload_type, int associated_payload_type);
@@ -212,6 +206,7 @@
uint16_t data_length_in_bytes);
// Returns statistics reported by the remote client in an RTCP packet.
+ // TODO(pbos): Remove this along with VideoSendStream::GetRtt().
int32_t GetSendRtcpStatistics(uint16_t* fraction_lost,
uint32_t* cumulative_lost,
uint32_t* extended_max,
@@ -222,23 +217,10 @@
void RegisterSendChannelRtcpStatisticsCallback(
RtcpStatisticsCallback* callback);
- // Returns our localy created statistics of the received RTP stream.
- int32_t GetReceivedRtcpStatistics(uint16_t* fraction_lost,
- uint32_t* cumulative_lost,
- uint32_t* extended_max,
- uint32_t* jitter_samples,
- int64_t* rtt_ms);
-
// Called on generation of RTCP stats
void RegisterReceiveChannelRtcpStatisticsCallback(
RtcpStatisticsCallback* callback);
- // Gets sent/received packets statistics.
- int32_t GetRtpStatistics(size_t* bytes_sent,
- uint32_t* packets_sent,
- size_t* bytes_received,
- uint32_t* packets_received) const;
-
// Gets send statistics for the rtp and rtx stream.
void GetSendStreamDataCounters(StreamDataCounters* rtp_counters,
StreamDataCounters* rtx_counters) const;
@@ -261,13 +243,6 @@
void GetReceiveRtcpPacketTypeCounter(
RtcpPacketTypeCounter* packet_counter) const;
- void GetBandwidthUsage(uint32_t* total_bitrate_sent,
- uint32_t* video_bitrate_sent,
- uint32_t* fec_bitrate_sent,
- uint32_t* nackBitrateSent) const;
- // TODO(holmer): Deprecated. We should use the SendSideDelayObserver instead
- // to avoid deadlocks.
- bool GetSendSideDelay(int* avg_send_delay, int* max_send_delay) const;
void RegisterSendSideDelayObserver(SendSideDelayObserver* observer);
// Called on any new send bitrate estimate.
@@ -292,27 +267,6 @@
const bool added);
virtual void ResetStatistics(uint32_t);
- int32_t SetLocalReceiver(const uint16_t rtp_port,
- const uint16_t rtcp_port,
- const char* ip_address);
- int32_t GetLocalReceiver(uint16_t* rtp_port,
- uint16_t* rtcp_port,
- char* ip_address) const;
- int32_t SetSendDestination(const char* ip_address,
- const uint16_t rtp_port,
- const uint16_t rtcp_port,
- const uint16_t source_rtp_port,
- const uint16_t source_rtcp_port);
- int32_t GetSendDestination(char* ip_address,
- uint16_t* rtp_port,
- uint16_t* rtcp_port,
- uint16_t* source_rtp_port,
- uint16_t* source_rtcp_port) const;
- int32_t GetSourceInfo(uint16_t* rtp_port,
- uint16_t* rtcp_port,
- char* ip_address,
- uint32_t ip_address_length);
-
int32_t SetRemoteSSRCType(const StreamType usage, const uint32_t SSRC);
int32_t StartSend();
@@ -586,7 +540,6 @@
I420FrameCallback* pre_render_callback_;
rtc::scoped_ptr<ReportBlockStats> report_block_stats_sender_;
- rtc::scoped_ptr<ReportBlockStats> report_block_stats_receiver_;
};
} // namespace webrtc
