Remove more unneeded things from ChannelSend

- SetNACKStatus() - only affects NetEq and RTP receiver
- GetRtpTimestampRateHz() - never used.
- ResendPackets() - never used.

Bug: webrtc:9801
Change-Id: I280b620723eb6917624f30f503eb8b8c88144e6d
Reviewed-on: https://webrtc-review.googlesource.com/c/111460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25721}

audio/audio_send_stream.cc

@@ -255,13 +255,6 @@ void AudioSendStream::ConfigureStream(
   if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {
     channel_send->SetRTCP_CNAME(new_config.rtp.c_name);
   }
-  // TODO(solenberg): Config NACK history window (which is a packet count),
-  // using the actual packet size for the configured codec.
-  if (first_time || old_config.rtp.nack.rtp_history_ms !=
-                        new_config.rtp.nack.rtp_history_ms) {
-    channel_send->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
-                                new_config.rtp.nack.rtp_history_ms / 20);
-  }
 
   if (first_time || new_config.send_transport != old_config.send_transport) {
     if (old_config.send_transport) {

audio/audio_send_stream_unittest.cc

@@ -151,7 +151,6 @@ struct ConfigHelper {
     stream_config_.send_codec_spec =
         AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
     stream_config_.rtp.ssrc = kSsrc;
-    stream_config_.rtp.nack.rtp_history_ms = 200;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
@@ -195,7 +194,6 @@ struct ConfigHelper {
     }));
     EXPECT_CALL(*channel_send_, SetLocalSSRC(kSsrc)).Times(1);
     EXPECT_CALL(*channel_send_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
-    EXPECT_CALL(*channel_send_, SetNACKStatus(true, 10)).Times(1);
     EXPECT_CALL(*channel_send_, SetFrameEncryptor(_)).Times(1);
     EXPECT_CALL(*channel_send_, SetExtmapAllowMixed(false)).Times(1);
     EXPECT_CALL(*channel_send_,
@@ -335,8 +333,8 @@ TEST(AudioSendStreamTest, ConfigToString) {
   config.rtcp_report_interval_ms = 2500;
   EXPECT_EQ(
       "{rtp: {ssrc: 1234, extmap-allow-mixed: true, extensions: [{uri: "
-      "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: "
+      "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], "
-      "{rtp_history_ms: 0}, c_name: foo_name}, rtcp_report_interval_ms: 2500, "
+      "c_name: foo_name}, rtcp_report_interval_ms: 2500, "
       "send_transport: null, media_transport: null, "
       "min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
       "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "

audio/channel_send.cc

@@ -146,7 +146,6 @@ class ChannelSend
   void SetRTCP_CNAME(absl::string_view c_name) override;
   std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const override;
   CallSendStatistics GetRTCPStatistics() const override;
-  void SetNACKStatus(bool enable, int max_packets) override;
 
   // ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
   // which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
@@ -199,15 +198,11 @@ class ChannelSend
   void OnUplinkPacketLossRate(float packet_loss_rate);
   bool InputMute() const;
 
-  int ResendPackets(const uint16_t* sequence_numbers, int length);
-
   int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type, int id);
 
   void UpdateOverheadForEncoder()
       RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
 
-  int GetRtpTimestampRateHz() const;
-
   int32_t SendRtpAudio(FrameType frameType,
                        uint8_t payloadType,
                        uint32_t timeStamp,
@@ -1174,20 +1169,6 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
   return stats;
 }
 
-void ChannelSend::SetNACKStatus(bool enable, int max_packets) {
-  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  // None of these functions can fail.
-  if (enable)
-    audio_coding_->EnableNack(max_packets);
-  else
-    audio_coding_->DisableNack();
-}
-
-// Called when we are missing one or more packets.
-int ChannelSend::ResendPackets(const uint16_t* sequence_numbers, int length) {
-  return _rtpRtcpModule->SendNACK(sequence_numbers, length);
-}
-
 void ChannelSend::ProcessAndEncodeAudio(
     std::unique_ptr<AudioFrame> audio_frame) {
   RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
@@ -1293,17 +1274,6 @@ int ChannelSend::SetSendRtpHeaderExtension(bool enable,
   return error;
 }
 
-int ChannelSend::GetRtpTimestampRateHz() const {
-  const auto format = audio_coding_->ReceiveFormat();
-  // Default to the playout frequency if we've not gotten any packets yet.
-  // TODO(ossu): Zero clockrate can only happen if we've added an external
-  // decoder for a format we don't support internally. Remove once that way of
-  // adding decoders is gone!
-  return (format && format->clockrate_hz != 0)
-             ? format->clockrate_hz
-             : audio_coding_->PlayoutFrequency();
-}
-
 int64_t ChannelSend::GetRTT() const {
   if (media_transport_) {
     // GetRTT is generally used in the RTCP codepath, where media transport is

audio/channel_send.h

@@ -59,7 +59,6 @@ class ChannelSendInterface {
   virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0;
 
   virtual CallSendStatistics GetRTCPStatistics() const = 0;
-  virtual void SetNACKStatus(bool enable, int max_packets) = 0;
 
   virtual bool SetEncoder(int payload_type,
                           std::unique_ptr<AudioEncoder> encoder) = 0;

audio/mock_voe_channel_proxy.h

@@ -75,7 +75,6 @@ class MockChannelSend : public voe::ChannelSendInterface {
   MOCK_METHOD2(SetMid, void(const std::string& mid, int extension_id));
   MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
   MOCK_METHOD1(SetRTCP_CNAME, void(absl::string_view c_name));
-  MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
   MOCK_METHOD1(SetExtmapAllowMixed, void(bool extmap_allow_mixed));
   MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
   MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));

call/audio_send_stream.cc

@@ -62,7 +62,6 @@ std::string AudioSendStream::Config::Rtp::ToString() const {
     }
   }
   ss << ']';
-  ss << ", nack: " << nack.ToString();
   ss << ", c_name: " << c_name;
   ss << '}';
   return ss.str();

call/audio_send_stream.h

@@ -89,9 +89,6 @@ class AudioSendStream {
       // RTP header extensions used for the sent stream.
       std::vector<RtpExtension> extensions;
 
-      // See NackConfig for description.
-      NackConfig nack;
-
       // RTCP CNAME, see RFC 3550.
       std::string c_name;
     } rtp;

media/engine/webrtcvoiceengine.cc

@@ -1024,8 +1024,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   void UpdateSendCodecSpec(
       const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    config_.rtp.nack.rtp_history_ms =
-        send_codec_spec.nack_enabled ? kNackRtpHistoryMs : 0;
     config_.send_codec_spec = send_codec_spec;
     auto info =
         config_.encoder_factory->QueryAudioEncoder(send_codec_spec.format);

media/engine/webrtcvoiceengine_unittest.cc

@@ -1635,18 +1635,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthForAudioDoesntAffectBwe) {
   SetSendParameters(send_parameters_);
 }
 
-// Test that we can enable NACK with opus as caller.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCaller) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
-      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
-  EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
-  SetSendParameters(parameters);
-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
-}
-
 // Test that we can enable NACK with opus as callee.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
   EXPECT_TRUE(SetupRecvStream());
@@ -1661,7 +1649,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
 
   EXPECT_TRUE(
       channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
 }
 
 // Test that we can enable NACK on receive streams.
@@ -1672,29 +1659,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) {
   parameters.codecs.push_back(kOpusCodec);
   parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
       cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
-  EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
   EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
   SetSendParameters(parameters);
-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
   EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
 }
 
-// Test that we can disable NACK.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNack) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
-      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
-  SetSendParameters(parameters);
-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
-
-  parameters.codecs.clear();
-  parameters.codecs.push_back(kOpusCodec);
-  SetSendParameters(parameters);
-  EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
-}
-
 // Test that we can disable NACK on receive streams.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) {
   EXPECT_TRUE(SetupSendStream());
@@ -1704,13 +1673,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) {
   parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
       cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
   SetSendParameters(parameters);
-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
   EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
 
   parameters.codecs.clear();
   parameters.codecs.push_back(kOpusCodec);
   SetSendParameters(parameters);
-  EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
   EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
 }
 
@@ -1723,7 +1690,6 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
   parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
       cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
   SetSendParameters(parameters);
-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
 
   EXPECT_TRUE(AddRecvStream(kSsrcY));
   EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);

video/end_to_end_tests/retransmission_tests.cc

@@ -165,7 +165,6 @@ TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
     void ModifyAudioConfigs(
         AudioSendStream::Config* send_config,
         std::vector<AudioReceiveStream::Config>* receive_configs) override {
-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
       remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;