WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

webrtc/modules/audio_coding/acm2/audio_coding_module.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.
  */
 
@@ skipping 103 matching lines @@
 
   // Initialize receiver, resets codec database etc.
   int InitializeReceiver() override;
 
   // Get current receive frequency.
   int ReceiveFrequency() const override;
 
   // 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;
 
   int RegisterReceiveCodec(const CodecInst& receive_codec) override;
   int RegisterReceiveCodec(
       const CodecInst& receive_codec,
       rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) override;
 
   int RegisterExternalReceiveCodec(int rtp_payload_type,
                                    AudioDecoder* external_decoder,
@@ skipping 177 matching lines @@
 };
 
 // Adds a codec usage sample to the histogram.
 void UpdateCodecTypeHistogram(size_t codec_type) {
   RTC_HISTOGRAM_ENUMERATION(
       "WebRTC.Audio.Encoder.CodecType", static_cast<int>(codec_type),
       static_cast<int>(
           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,
             int16_t* out_buff) {
   if (length_out_buff < frame.samples_per_channel_) {
     return -1;
   }
   for (size_t n = 0; n < frame.samples_per_channel_; ++n)
     out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1;
   return 0;
@@ skipping 608 matching lines @@
   // If the receiver is already initialized then we want to destroy any
   // existing decoders. After a call to this function, we should have a clean
   // start-up.
   if (receiver_initialized_)
     receiver_.RemoveAllCodecs();
   receiver_.ResetInitialDelay();
   receiver_.SetMinimumDelay(0);
   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;
 }
 
 // Get current receive frequency.
 int AudioCodingModuleImpl::ReceiveFrequency() const {
   const auto last_packet_sample_rate = receiver_.last_packet_sample_rate_hz();
   return last_packet_sample_rate ? *last_packet_sample_rate
                                  : receiver_.last_output_sample_rate_hz();
 }
 
 // Get current playout frequency.
 int AudioCodingModuleImpl::PlayoutFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "PlayoutFrequency()");
   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) {
   rtc::CritScope lock(&acm_crit_sect_);
   RTC_DCHECK(receiver_initialized_);
 
   if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
     LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
                     << " for decoder.";
     return false;
@@ skipping 376 matching lines @@
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   bool valid = acm2::RentACodec::IsCodecValid(codec);
   if (!valid)
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
                  "Invalid codec setting");
   return valid;
 }
 
 }  // namespace webrtc