Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )

Reason for revert:
Makes perf and Chromium FYI bots unhappy.

Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdba

TBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}

webrtc/audio/audio_receive_stream.cc

@@ -94,7 +94,10 @@ AudioReceiveStream::AudioReceiveStream(
                channel_proxy_->GetAudioDecoderFactory());
 
   channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
-  channel_proxy_->SetReceiveCodecs(config.decoder_map);
+
+  for (const auto& kv : config.decoder_map) {
+    channel_proxy_->SetRecPayloadType(kv.first, kv.second);
+  }
 
   for (const auto& extension : config.rtp.extensions) {
     if (extension.uri == RtpExtension::kAudioLevelUri) {

webrtc/audio/audio_receive_stream_unittest.cc

@@ -8,7 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <map>
 #include <string>
 #include <vector>
 
@@ -112,11 +111,6 @@ struct ConfigHelper {
               .Times(1)
               .After(expect_set);
           EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
-          EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_))
-              .WillRepeatedly(
-                  Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
-                    EXPECT_THAT(codecs, testing::IsEmpty());
-                  }));
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;

webrtc/call/call_unittest.cc

@@ -9,7 +9,6 @@
  */
 
 #include <list>
-#include <map>
 #include <memory>
 
 #include "webrtc/call/audio_state.h"
@@ -142,11 +141,6 @@ TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
             new testing::NiceMock<test::MockVoEChannelProxy>();
         EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
             .WillRepeatedly(testing::ReturnRef(decoder_factory));
-        EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_))
-            .WillRepeatedly(testing::Invoke(
-                [](const std::map<int, SdpAudioFormat>& codecs) {
-                  EXPECT_THAT(codecs, testing::IsEmpty());
-                }));
         // If being called for the send channel, save a pointer to the channel
         // proxy for later.
         if (channel_id == kRecvChannelId) {
@@ -194,11 +188,6 @@ TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
             new testing::NiceMock<test::MockVoEChannelProxy>();
         EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
             .WillRepeatedly(testing::ReturnRef(decoder_factory));
-        EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_))
-            .WillRepeatedly(testing::Invoke(
-                [](const std::map<int, SdpAudioFormat>& codecs) {
-                  EXPECT_THAT(codecs, testing::IsEmpty());
-                }));
         // If being called for the send channel, save a pointer to the channel
         // proxy for later.
         if (channel_id == kRecvChannelId) {

webrtc/media/engine/fakewebrtcvoiceengine.h

@@ -117,6 +117,8 @@ class FakeWebRtcVoiceEngine
       return -1;
     }
     Channel* ch = new Channel();
+    auto db = webrtc::acm2::RentACodec::Database();
+    ch->recv_codecs.assign(db.begin(), db.end());
     ch->neteq_capacity = config.acm_config.neteq_config.max_packets_in_buffer;
     ch->neteq_fast_accelerate =
         config.acm_config.neteq_config.enable_fast_accelerate;

webrtc/media/engine/webrtcvoiceengine.cc

@@ -132,6 +132,13 @@ std::string ToString(const AudioCodec& codec) {
   return ss.str();
 }
 
+std::string ToString(const webrtc::CodecInst& codec) {
+  std::stringstream ss;
+  ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
+     << " (" << codec.pltype << ")";
+  return ss.str();
+}
+
 bool IsCodec(const AudioCodec& codec, const char* ref_name) {
   return (_stricmp(codec.name.c_str(), ref_name) == 0);
 }
@@ -1459,8 +1466,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
       const std::vector<webrtc::RtpExtension>& extensions,
       webrtc::Call* call,
       webrtc::Transport* rtcp_send_transport,
-      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
+      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
-      const std::map<int, webrtc::SdpAudioFormat>& decoder_map)
       : call_(call), config_() {
     RTC_DCHECK_GE(ch, 0);
     RTC_DCHECK(call);
@@ -1473,7 +1479,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
     config_.voe_channel_id = ch;
     config_.sync_group = sync_group;
     config_.decoder_factory = decoder_factory;
-    config_.decoder_map = decoder_map;
     RecreateAudioReceiveStream();
   }
 
@@ -1862,9 +1867,8 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
     ChangePlayout(false);
   }
 
-  decoder_map_ = std::move(decoder_map);
   for (auto& kv : recv_streams_) {
-    kv.second->RecreateAudioReceiveStream(decoder_map_);
+    kv.second->RecreateAudioReceiveStream(decoder_map);
   }
   recv_codecs_ = codecs;
 
@@ -2221,12 +2225,38 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
     return false;
   }
 
+  // Turn off all supported codecs.
+  // TODO(solenberg): Remove once "no codecs" is the default state of a stream.
+  for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
+    voe_codec.pltype = -1;
+    if (engine()->voe()->codec()->SetRecPayloadType(channel, voe_codec) == -1) {
+      LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+      DeleteVoEChannel(channel);
+      return false;
+    }
+  }
+
+  // Only enable those configured for this channel.
+  for (const auto& codec : recv_codecs_) {
+    webrtc::CodecInst voe_codec = {0};
+    if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
+      voe_codec.pltype = codec.id;
+      if (engine()->voe()->codec()->SetRecPayloadType(
+          channel, voe_codec) == -1) {
+        LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+        DeleteVoEChannel(channel);
+        return false;
+      }
+    }
+  }
+
   recv_streams_.insert(std::make_pair(
-      ssrc,
+      ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
-      new WebRtcAudioReceiveStream(
+                                         recv_transport_cc_enabled_,
-          channel, ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
+                                         recv_nack_enabled_,
-          recv_nack_enabled_, sp.sync_label, recv_rtp_extensions_, call_, this,
+                                         sp.sync_label, recv_rtp_extensions_,
-          engine()->decoder_factory_, decoder_map_)));
+                                         call_, this,
+                                         engine()->decoder_factory_)));
   recv_streams_[ssrc]->SetPlayout(playout_);
 
   return true;

webrtc/media/engine/webrtcvoiceengine.h

@@ -251,12 +251,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> send_codecs_;
-
-  // TODO(kwiberg): decoder_map_ and recv_codecs_ store the exact same
-  // information, in slightly different formats. Eliminate recv_codecs_.
-  std::map<int, webrtc::SdpAudioFormat> decoder_map_;
   std::vector<AudioCodec> recv_codecs_;
-
   int max_send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;

webrtc/media/engine/webrtcvoiceengine_unittest.cc

@@ -31,7 +31,6 @@
 #include "webrtc/test/gtest.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 
-using testing::ContainerEq;
 using testing::Return;
 using testing::StrictMock;
 
@@ -796,12 +795,26 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
   parameters.codecs[2].id = 126;
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+  int channel_num = voe_.GetLastChannel();
-              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+
-                  {{0, {"PCMU", 8000, 1}},
+  webrtc::CodecInst gcodec;
-                   {106, {"ISAC", 16000, 1}},
+  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
-                   {126, {"telephone-event", 8000, 1}},
+  gcodec.plfreq = 16000;
-                   {107, {"telephone-event", 32000, 1}}})));
+  gcodec.channels = 1;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
+  EXPECT_EQ(106, gcodec.pltype);
+  EXPECT_STREQ("ISAC", gcodec.plname);
+
+  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
+  gcodec.plfreq = 8000;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
+  EXPECT_EQ(126, gcodec.pltype);
+  EXPECT_STREQ("telephone-event", gcodec.plname);
+
+  gcodec.plfreq = 32000;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
+  EXPECT_EQ(107, gcodec.pltype);
+  EXPECT_STREQ("telephone-event", gcodec.plname);
 }
 
 // Test that we fail to set an unknown inbound codec.
@@ -832,11 +845,16 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) {
   parameters.codecs.push_back(kOpusCodec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+  int channel_num = voe_.GetLastChannel();
-              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+  webrtc::CodecInst opus;
-                  {{0, {"PCMU", 8000, 1}},
+  cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
-                   {103, {"ISAC", 16000, 1}},
+  // Even without stereo parameters, recv codecs still specify channels = 2.
-                   {111, {"opus", 48000, 2}}})));
+  EXPECT_EQ(2, opus.channels);
+  EXPECT_EQ(111, opus.pltype);
+  EXPECT_STREQ("opus", opus.plname);
+  opus.pltype = 0;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, opus));
+  EXPECT_EQ(111, opus.pltype);
 }
 
 // Test that we can decode OPUS with stereo = 0.
@@ -849,11 +867,16 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) {
   parameters.codecs[2].params["stereo"] = "0";
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+  int channel_num2 = voe_.GetLastChannel();
-              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+  webrtc::CodecInst opus;
-                  {{0, {"PCMU", 8000, 1}},
+  cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
-                   {103, {"ISAC", 16000, 1}},
+  // Even when stereo is off, recv codecs still specify channels = 2.
-                   {111, {"opus", 48000, 2, {{"stereo", "0"}}}}})));
+  EXPECT_EQ(2, opus.channels);
+  EXPECT_EQ(111, opus.pltype);
+  EXPECT_STREQ("opus", opus.plname);
+  opus.pltype = 0;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus));
+  EXPECT_EQ(111, opus.pltype);
 }
 
 // Test that we can decode OPUS with stereo = 1.
@@ -866,11 +889,15 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) {
   parameters.codecs[2].params["stereo"] = "1";
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+  int channel_num2 = voe_.GetLastChannel();
-              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+  webrtc::CodecInst opus;
-                  {{0, {"PCMU", 8000, 1}},
+  cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
-                   {103, {"ISAC", 16000, 1}},
+  EXPECT_EQ(2, opus.channels);
-                   {111, {"opus", 48000, 2, {{"stereo", "1"}}}}})));
+  EXPECT_EQ(111, opus.pltype);
+  EXPECT_STREQ("opus", opus.plname);
+  opus.pltype = 0;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus));
+  EXPECT_EQ(111, opus.pltype);
 }
 
 // Test that changes to recv codecs are applied to all streams.
@@ -884,15 +911,28 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
   parameters.codecs[0].id = 106;  // collide with existing CN 32k
   parameters.codecs[2].id = 126;
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
-  for (const auto& ssrc : {kSsrcX, kSsrcY}) {
+  EXPECT_TRUE(AddRecvStream(kSsrcX));
-    EXPECT_TRUE(AddRecvStream(ssrc));
+  int channel_num2 = voe_.GetLastChannel();
-    EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map,
+
-                (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+  webrtc::CodecInst gcodec;
-                    {{0, {"PCMU", 8000, 1}},
+  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
-                     {106, {"ISAC", 16000, 1}},
+  gcodec.plfreq = 16000;
-                     {126, {"telephone-event", 8000, 1}},
+  gcodec.channels = 1;
-                     {107, {"telephone-event", 32000, 1}}})));
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
-  }
+  EXPECT_EQ(106, gcodec.pltype);
+  EXPECT_STREQ("ISAC", gcodec.plname);
+
+  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
+  gcodec.plfreq = 8000;
+  gcodec.channels = 1;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
+  EXPECT_EQ(126, gcodec.pltype);
+  EXPECT_STREQ("telephone-event", gcodec.plname);
+
+  gcodec.plfreq = 32000;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
+  EXPECT_EQ(107, gcodec.pltype);
+  EXPECT_STREQ("telephone-event", gcodec.plname);
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
@@ -2921,9 +2961,12 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) {
   parameters.codecs.push_back(kPcmuCodec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+  int channel_num2 = voe_.GetLastChannel();
-              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+  webrtc::CodecInst gcodec;
-                  {{0, {"PCMU", 8000, 1}}, {103, {"ISAC", 16000, 1}}})));
+  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "opus");
+  gcodec.plfreq = 48000;
+  gcodec.channels = 2;
+  EXPECT_EQ(-1, voe_.GetRecPayloadType(channel_num2, gcodec));
 }
 
 // Test that we properly clean up any streams that were added, even if

webrtc/modules/audio_coding/acm2/acm_receiver.cc

@@ -179,10 +179,6 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
   return 0;
 }
 
-void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
-  neteq_->SetCodecs(codecs);
-}
-
 int32_t AcmReceiver::AddCodec(int acm_codec_id,
                               uint8_t payload_type,
                               size_t channels,

webrtc/modules/audio_coding/acm2/acm_receiver.h

@@ -79,9 +79,6 @@ class AcmReceiver {
   //
   int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
 
-  // Replace the current set of decoders with the specified set.
-  void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
-
   //
   // Adds a new codec to the NetEq codec database.
   //

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

@@ -121,8 +121,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // Get current playout frequency.
   int PlayoutFrequency() const override;
 
-  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
-
   bool RegisterReceiveCodec(int rtp_payload_type,
                             const SdpAudioFormat& audio_format) override;
 
@@ -320,6 +318,16 @@ void UpdateCodecTypeHistogram(size_t codec_type) {
           webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
 }
 
+// TODO(turajs): the same functionality is used in NetEq. If both classes
+// need them, make it a static function in ACMCodecDB.
+bool IsCodecRED(const CodecInst& codec) {
+  return (STR_CASE_CMP(codec.plname, "RED") == 0);
+}
+
+bool IsCodecCN(const CodecInst& codec) {
+  return (STR_CASE_CMP(codec.plname, "CN") == 0);
+}
+
 // Stereo-to-mono can be used as in-place.
 int DownMix(const AudioFrame& frame,
             size_t length_out_buff,
@@ -948,6 +956,19 @@ int AudioCodingModuleImpl::InitializeReceiverSafe() {
   receiver_.SetMaximumDelay(0);
   receiver_.FlushBuffers();
 
+  // Register RED and CN.
+  auto db = acm2::RentACodec::Database();
+  for (size_t i = 0; i < db.size(); i++) {
+    if (IsCodecRED(db[i]) || IsCodecCN(db[i])) {
+      if (receiver_.AddCodec(static_cast<int>(i),
+                             static_cast<uint8_t>(db[i].pltype), 1,
+                             db[i].plfreq, nullptr, db[i].plname) < 0) {
+        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                     "Cannot register master codec.");
+        return -1;
+      }
+    }
+  }
   receiver_initialized_ = true;
   return 0;
 }
@@ -966,12 +987,6 @@ int AudioCodingModuleImpl::PlayoutFrequency() const {
   return receiver_.last_output_sample_rate_hz();
 }
 
-void AudioCodingModuleImpl::SetReceiveCodecs(
-    const std::map<int, SdpAudioFormat>& codecs) {
-  rtc::CritScope lock(&acm_crit_sect_);
-  receiver_.SetCodecs(codecs);
-}
-
 bool AudioCodingModuleImpl::RegisterReceiveCodec(
     int rtp_payload_type,
     const SdpAudioFormat& audio_format) {

webrtc/modules/audio_coding/include/audio_coding_module.h

@@ -485,10 +485,6 @@ class AudioCodingModule {
   //
   virtual int32_t PlayoutFrequency() const = 0;
 
-  // Replace any existing decoders with the given payload type -> decoder map.
-  virtual void SetReceiveCodecs(
-      const std::map<int, SdpAudioFormat>& codecs) = 0;
-
   // Registers a decoder for the given payload type. Returns true iff
   // successful.
   virtual bool RegisterReceiveCodec(int rtp_payload_type,

webrtc/modules/audio_coding/neteq/decoder_database.cc

@@ -123,38 +123,6 @@ void DecoderDatabase::Reset() {
   active_cng_decoder_type_ = -1;
 }
 
-std::vector<int> DecoderDatabase::SetCodecs(
-    const std::map<int, SdpAudioFormat>& codecs) {
-  // First collect all payload types that we'll remove or reassign, then remove
-  // them from the database.
-  std::vector<int> changed_payload_types;
-  for (const std::pair<uint8_t, const DecoderInfo&> kv : decoders_) {
-    auto i = codecs.find(kv.first);
-    if (i == codecs.end() || i->second != kv.second.GetFormat()) {
-      changed_payload_types.push_back(kv.first);
-    }
-  }
-  for (int pl_type : changed_payload_types) {
-    Remove(pl_type);
-  }
-
-  // Enter the new and changed payload type mappings into the database.
-  for (const auto& kv : codecs) {
-    const int& rtp_payload_type = kv.first;
-    const SdpAudioFormat& audio_format = kv.second;
-    RTC_DCHECK_GE(rtp_payload_type, 0);
-    RTC_DCHECK_LE(rtp_payload_type, 0x7f);
-    if (decoders_.count(rtp_payload_type) == 0) {
-      decoders_.insert(std::make_pair(
-          rtp_payload_type, DecoderInfo(audio_format, decoder_factory_.get())));
-    } else {
-      // The mapping for this payload type hasn't changed.
-    }
-  }
-
-  return changed_payload_types;
-}
-
 int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
                                      NetEqDecoder codec_type,
                                      const std::string& name) {

webrtc/modules/audio_coding/neteq/decoder_database.h

@@ -149,11 +149,6 @@ class DecoderDatabase {
   // using InsertExternal().
   virtual void Reset();
 
-  // Replaces the existing set of decoders with the given set. Returns the
-  // payload types that were reassigned or removed while doing so.
-  virtual std::vector<int> SetCodecs(
-      const std::map<int, SdpAudioFormat>& codecs);
-
   // Registers |rtp_payload_type| as a decoder of type |codec_type|. The |name|
   // is only used to populate the name field in the DecoderInfo struct in the
   // database, and can be arbitrary (including empty). Returns kOK on success;

webrtc/modules/audio_coding/neteq/include/neteq.h

@@ -157,9 +157,6 @@ class NetEq {
   // Returns kOK on success, or kFail in case of an error.
   virtual int GetAudio(AudioFrame* audio_frame, bool* muted) = 0;
 
-  // Replaces the current set of decoders with the given one.
-  virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;
-
   // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
   // information in the codec database. Returns 0 on success, -1 on failure.
   // The name is only used to provide information back to the caller about the

webrtc/modules/audio_coding/neteq/neteq_impl.cc

@@ -212,15 +212,6 @@ int NetEqImpl::GetAudio(AudioFrame* audio_frame, bool* muted) {
   return kOK;
 }
 
-void NetEqImpl::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
-  rtc::CritScope lock(&crit_sect_);
-  const std::vector<int> changed_payload_types =
-      decoder_database_->SetCodecs(codecs);
-  for (const int pt : changed_payload_types) {
-    packet_buffer_->DiscardPacketsWithPayloadType(pt);
-  }
-}
-
 int NetEqImpl::RegisterPayloadType(NetEqDecoder codec,
                                    const std::string& name,
                                    uint8_t rtp_payload_type) {

webrtc/modules/audio_coding/neteq/neteq_impl.h

@@ -111,8 +111,6 @@ class NetEqImpl : public webrtc::NetEq {
 
   int GetAudio(AudioFrame* audio_frame, bool* muted) override;
 
-  void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
-
   int RegisterPayloadType(NetEqDecoder codec,
                           const std::string& codec_name,
                           uint8_t rtp_payload_type) override;

webrtc/modules/rtp_rtcp/BUILD.gn

@@ -168,7 +168,6 @@ rtc_static_library("rtp_rtcp") {
   deps = [
     "../..:webrtc_common",
     "../../api:transport_api",
-    "../../api/audio_codecs:audio_codecs_api",
     "../../base:gtest_prod",
     "../../base:rtc_base_approved",
     "../../base:rtc_task_queue",
@@ -176,7 +175,6 @@ rtc_static_library("rtp_rtcp") {
     "../../common_video",
     "../../logging:rtc_event_log_api",
     "../../system_wrappers",
-    "../audio_coding:audio_format_conversion",
     "../remote_bitrate_estimator",
   ]
 

webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h

@@ -15,7 +15,6 @@
 #include <memory>
 #include <set>
 
-#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/deprecation.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -44,10 +43,6 @@ class RTPPayloadRegistry {
 
   // TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
   // and simplify the code. http://crbug/webrtc/6743.
-
-  // Replace all audio receive payload types with the given map.
-  void SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs);
-
   int32_t RegisterReceivePayload(const CodecInst& audio_codec,
                                  bool* created_new_payload_type);
   int32_t RegisterReceivePayload(const VideoCodec& video_codec);

webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc

@@ -16,7 +16,6 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 
 namespace webrtc {
@@ -120,31 +119,6 @@ RTPPayloadRegistry::RTPPayloadRegistry()
 
 RTPPayloadRegistry::~RTPPayloadRegistry() = default;
 
-void RTPPayloadRegistry::SetAudioReceivePayloads(
-    std::map<int, SdpAudioFormat> codecs) {
-  rtc::CritScope cs(&crit_sect_);
-
-#if RTC_DCHECK_IS_ON
-  RTC_DCHECK(!used_for_video_);
-  used_for_audio_ = true;
-#endif
-
-  payload_type_map_.clear();
-  for (const auto& kv : codecs) {
-    const int& rtp_payload_type = kv.first;
-    const SdpAudioFormat& audio_format = kv.second;
-    const CodecInst ci = SdpToCodecInst(rtp_payload_type, audio_format);
-    RTC_DCHECK(IsPayloadTypeValid(rtp_payload_type));
-    payload_type_map_.insert(
-        std::make_pair(rtp_payload_type, CreatePayloadType(ci)));
-  }
-
-  // Clear the value of last received payload type since it might mean
-  // something else now.
-  last_received_payload_type_ = -1;
-  last_received_media_payload_type_ = -1;
-}
-
 int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec,
                                                    bool* created_new_payload) {
   rtc::CritScope cs(&crit_sect_);

webrtc/test/mock_voe_channel_proxy.h

@@ -87,8 +87,6 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   MOCK_METHOD1(SetSendCodec, bool(const CodecInst& codec_inst));
   MOCK_METHOD2(SetSendCNPayloadType,
                bool(int type, PayloadFrequencies frequency));
-  MOCK_METHOD1(SetReceiveCodecs,
-               void(const std::map<int, SdpAudioFormat>& codecs));
   MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
   MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
                void(float recoverable_packet_loss_rate));

webrtc/test/mock_voice_engine.h

@@ -52,11 +52,6 @@ class MockVoiceEngine : public VoiceEngineImpl {
               new testing::NiceMock<webrtc::test::MockVoEChannelProxy>();
           EXPECT_CALL(*proxy, GetAudioDecoderFactory())
               .WillRepeatedly(testing::ReturnRef(decoder_factory_));
-          EXPECT_CALL(*proxy, SetReceiveCodecs(testing::_))
-              .WillRepeatedly(testing::Invoke(
-                  [](const std::map<int, SdpAudioFormat>& codecs) {
-                    EXPECT_THAT(codecs, testing::IsEmpty());
-                  }));
           return proxy;
         }));
 

webrtc/voice_engine/channel.cc

@@ -987,10 +987,9 @@ int32_t Channel::Init() {
     return -1;
   }
 
-  return 0;
+  // --- Register all supported codecs to the receiving side of the
-}
+  // RTP/RTCP module
 
-void Channel::RegisterLegacyCodecs() {
   CodecInst codec;
   const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
 
@@ -1042,6 +1041,8 @@ void Channel::RegisterLegacyCodecs() {
       }
     }
   }
+
+  return 0;
 }
 
 void Channel::Terminate() {
@@ -1359,11 +1360,6 @@ int32_t Channel::GetVADStatus(bool& enabledVAD,
   return 0;
 }
 
-void Channel::SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
-  rtp_payload_registry_->SetAudioReceivePayloads(codecs);
-  audio_coding_->SetReceiveCodecs(codecs);
-}
-
 int32_t Channel::SetRecPayloadType(const CodecInst& codec) {
   return SetRecPayloadType(codec.pltype, CodecInstToSdp(codec));
 }

webrtc/voice_engine/channel.h

@@ -151,7 +151,6 @@ class Channel
           uint32_t instanceId,
           const VoEBase::ChannelConfig& config);
   int32_t Init();
-  void RegisterLegacyCodecs();
   void Terminate();
   int32_t SetEngineInformation(Statistics& engineStatistics,
                                OutputMixer& outputMixer,
@@ -169,8 +168,6 @@ class Channel
   // go.
   const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
 
-  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
-
   // API methods
 
   // VoEBase

webrtc/voice_engine/channel_proxy.cc

@@ -173,12 +173,6 @@ void ChannelProxy::SetRecPayloadType(int payload_type,
   RTC_DCHECK_EQ(0, result);
 }
 
-void ChannelProxy::SetReceiveCodecs(
-    const std::map<int, SdpAudioFormat>& codecs) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  channel()->SetReceiveCodecs(codecs);
-}
-
 void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   channel()->SetSink(std::move(sink));
@@ -385,11 +379,6 @@ void ChannelProxy::OnRecoverableUplinkPacketLossRate(
   channel()->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate);
 }
 
-void ChannelProxy::RegisterLegacyCodecs() {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  channel()->RegisterLegacyCodecs();
-}
-
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();

webrtc/voice_engine/channel_proxy.h

@@ -82,7 +82,6 @@ class ChannelProxy {
   virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
   virtual void SetRecPayloadType(int payload_type,
                                  const SdpAudioFormat& format);
-  virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
   virtual void SetInputMute(bool muted);
   virtual void RegisterExternalTransport(Transport* transport);
@@ -120,7 +119,6 @@ class ChannelProxy {
   virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
   virtual void OnRecoverableUplinkPacketLossRate(
       float recoverable_packet_loss_rate);
-  virtual void RegisterLegacyCodecs();
 
  private:
   Channel* channel() const;

webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

@@ -15,8 +15,6 @@
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/voice_engine/channel_proxy.h"
-#include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace {
   static const unsigned int kReflectorSsrc = 0x0000;
@@ -64,9 +62,6 @@ ConferenceTransport::ConferenceTransport()
 
   EXPECT_EQ(0, local_base_->Init());
   local_sender_ = local_base_->CreateChannel();
-  static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
-      ->GetChannelProxy(local_sender_)
-      ->RegisterLegacyCodecs();
   EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
   EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
   EXPECT_EQ(0, local_rtp_rtcp_->
@@ -77,9 +72,6 @@ ConferenceTransport::ConferenceTransport()
 
   EXPECT_EQ(0, remote_base_->Init());
   reflector_ = remote_base_->CreateChannel();
-  static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
-      ->GetChannelProxy(reflector_)
-      ->RegisterLegacyCodecs();
   EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
   EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
 
@@ -230,9 +222,6 @@ void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
 unsigned int ConferenceTransport::AddStream(std::string file_name,
                                             webrtc::FileFormats format) {
   const int new_sender = remote_base_->CreateChannel();
-  static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
-      ->GetChannelProxy(new_sender)
-      ->RegisterLegacyCodecs();
   EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));
 
   const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
@@ -246,9 +235,6 @@ unsigned int ConferenceTransport::AddStream(std::string file_name,
       new_sender, file_name.c_str(), true, false, format, 1.0));
 
   const int new_receiver = local_base_->CreateChannel();
-  static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
-      ->GetChannelProxy(new_receiver)
-      ->RegisterLegacyCodecs();
   EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));
 
   EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));

webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc

@@ -16,6 +16,5 @@ AfterStreamingFixture::AfterStreamingFixture()
   webrtc::VoiceEngineImpl* voe_impl =
       static_cast<webrtc::VoiceEngineImpl*>(voice_engine_);
   channel_proxy_ = voe_impl->GetChannelProxy(channel_);
-  channel_proxy_->RegisterLegacyCodecs();
   ResumePlaying();
 }

webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc

@@ -8,9 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
-#include "webrtc/voice_engine/voice_engine_impl.h"
 
 class CodecBeforeStreamingTest : public AfterInitializationFixture {
  protected:
@@ -21,9 +19,6 @@ class CodecBeforeStreamingTest : public AfterInitializationFixture {
     codec_instance_.pacsize = 480;
 
     channel_ = voe_base_->CreateChannel();
-    static_cast<webrtc::VoiceEngineImpl*>(voice_engine_)
-        ->GetChannelProxy(channel_)
-        ->RegisterLegacyCodecs();
   }
 
   void TearDown() {