Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/voice_engine/channel.h"
 
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/output_mixer.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
 namespace voe {
 
 namespace {
 
 constexpr int64_t kMaxRetransmissionWindowMs = 1000;
 constexpr int64_t kMinRetransmissionWindowMs = 30;
 
-bool RegisterReceiveCodec(std::unique_ptr<AudioCodingModule>* acm,
-                          acm2::RentACodec* rac,
-                          const CodecInst& ci) {
-  const int result = (*acm)->RegisterReceiveCodec(
-      ci, [&] { return rac->RentIsacDecoder(ci.plfreq); });
-  return result == 0;
-}
-
 }  // namespace
 
 const int kTelephoneEventAttenuationdB = 10;
 
 class RtcEventLogProxy final : public webrtc::RtcEventLog {
  public:
   RtcEventLogProxy() : event_log_(nullptr) {}
 
   bool StartLogging(const std::string& file_name,
                     int64_t max_size_bytes) override {
@@ skipping 422 matching lines @@
   receiveCodec.pltype = payloadType;
   receiveCodec.plfreq = frequency;
   receiveCodec.channels = channels;
   receiveCodec.rate = rate;
   strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
 
   audio_coding_->Codec(payloadName, &dummyCodec, frequency, channels);
   receiveCodec.pacsize = dummyCodec.pacsize;
 
   // Register the new codec to the ACM
-  if (!RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, receiveCodec)) {
+  if (!audio_coding_->RegisterReceiveCodec(receiveCodec.pltype,
+                                           CodecInstToSdp(receiveCodec))) {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::OnInitializeDecoder() invalid codec ("
                  "pt=%d, name=%s) received - 1",
                  payloadType, payloadName);
     _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR);
     return -1;
   }
 
   return 0;
 }
@@ skipping 547 matching lines @@
     }
 
     // Ensure that PCMU is used as default codec on the sending side
     if (!STR_CASE_CMP(codec.plname, "PCMU") && (codec.channels == 1)) {
       SetSendCodec(codec);
     }
 
     // Register default PT for outband 'telephone-event'
     if (!STR_CASE_CMP(codec.plname, "telephone-event")) {
       if (_rtpRtcpModule->RegisterSendPayload(codec) == -1 ||
-          !RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, codec)) {
+          !audio_coding_->RegisterReceiveCodec(codec.pltype,
+                                               CodecInstToSdp(codec))) {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                      "Channel::Init() failed to register outband "
                      "'telephone-event' (%d/%d) correctly",
                      codec.pltype, codec.plfreq);
       }
     }
 
     if (!STR_CASE_CMP(codec.plname, "CN")) {
       if (!codec_manager_.RegisterEncoder(codec) ||
           !codec_manager_.MakeEncoder(&rent_a_codec_, audio_coding_.get()) ||
-          !RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, codec) ||
+          !audio_coding_->RegisterReceiveCodec(codec.pltype,
+                                               CodecInstToSdp(codec)) ||
           _rtpRtcpModule->RegisterSendPayload(codec) == -1) {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                      "Channel::Init() failed to register CN (%d/%d) "
                      "correctly - 1",
                      codec.pltype, codec.plfreq);
       }
     }
   }
 
   return 0;
@@ skipping 327 matching lines @@
     rtp_receiver_->DeRegisterReceivePayload(codec.pltype);
     if (rtp_receiver_->RegisterReceivePayload(
             codec.plname, codec.pltype, codec.plfreq, codec.channels,
             (codec.rate < 0) ? 0 : codec.rate) != 0) {
       _engineStatisticsPtr->SetLastError(
           VE_RTP_RTCP_MODULE_ERROR, kTraceError,
           "SetRecPayloadType() RTP/RTCP-module registration failed");
       return -1;
     }
   }
-  if (!RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, codec)) {
+  if (!audio_coding_->RegisterReceiveCodec(codec.pltype,
+                                           CodecInstToSdp(codec))) {
     audio_coding_->UnregisterReceiveCodec(codec.pltype);
-    if (!RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, codec)) {
+    if (!audio_coding_->RegisterReceiveCodec(codec.pltype,
+                                             CodecInstToSdp(codec))) {
       _engineStatisticsPtr->SetLastError(
           VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
           "SetRecPayloadType() ACM registration failed - 1");
       return -1;
     }
   }
   return 0;
 }
 
 int32_t Channel::GetRecPayloadType(CodecInst& codec) {
@@ skipping 1826 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc