WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 15 matching lines @@
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
+#include "webrtc/media/engine/payload_type_mapper.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace cricket {
 namespace {
 
@@ skipping 196 matching lines @@
 webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) {
   webrtc::AudioState::Config config;
   config.voice_engine = voe_wrapper->engine();
   return config;
 }
 
 class WebRtcVoiceCodecs final {
  public:
   // TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec
   // list and add a test which verifies VoE supports the listed codecs.
-  static std::vector<AudioCodec> SupportedCodecs() {
+  static std::vector<AudioCodec> SupportedSendCodecs() {
     std::vector<AudioCodec> result;
     // Iterate first over our preferred codecs list, so that the results are
     // added in order of preference.
     for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
       const CodecPref* pref = &kCodecPrefs[i];
       for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
         // Change the sample rate of G722 to 8000 to match SDP.
         MaybeFixupG722(&voe_codec, 8000);
         // Skip uncompressed formats.
         if (IsCodec(voe_codec, kL16CodecName)) {
@@ skipping 242 matching lines @@
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(
     webrtc::AudioDeviceModule* adm,
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     VoEWrapper* voe_wrapper)
     : adm_(adm), decoder_factory_(decoder_factory), voe_wrapper_(voe_wrapper) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
   RTC_DCHECK(voe_wrapper);
+  RTC_DCHECK(decoder_factory);
 
   signal_thread_checker_.DetachFromThread();
 
   // Load our audio codec list.
-  LOG(LS_INFO) << "Supported codecs in order of preference:";
-  codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
-  for (const AudioCodec& codec : codecs_) {
+  LOG(LS_INFO) << "Supported send codecs in order of preference:";
+  send_codecs_ = WebRtcVoiceCodecs::SupportedSendCodecs();
+  for (const AudioCodec& codec : send_codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
 
+  LOG(LS_INFO) << "Supported recv codecs in order of preference:";
+  recv_codecs_ = CollectRecvCodecs();
+  for (const AudioCodec& codec : recv_codecs_) {
+    LOG(LS_INFO) << ToString(codec);
+  }
+
   voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
 
   // Temporarily turn logging level up for the Init() call.
   webrtc::Trace::SetTraceCallback(this);
   webrtc::Trace::set_level_filter(kElevatedTraceFilter);
   LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
   RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm_.get(), nullptr,
                                              decoder_factory_));
   webrtc::Trace::set_level_filter(kDefaultTraceFilter);
 
@@ skipping 364 matching lines @@
 
 int WebRtcVoiceEngine::GetInputLevel() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   unsigned int ulevel;
   return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
       static_cast<int>(ulevel) : -1;
 }
 
 const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
-  return codecs_;
+  return send_codecs_;
 }
 
 const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
-  return codecs_;
+  return recv_codecs_;
 }
 
 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
   RtpCapabilities capabilities;
   capabilities.header_extensions.push_back(
       webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
                            webrtc::RtpExtension::kAudioLevelDefaultId));
   capabilities.header_extensions.push_back(
       webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri,
@@ skipping 140 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return voe_wrapper_->base()->CreateChannel(voe_config_);
 }
 
 webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(adm_);
   return adm_;
 }
 
+AudioCodecs WebRtcVoiceEngine::CollectRecvCodecs() const {
+  PayloadTypeMapper mapper;
+  AudioCodecs out;
+  const std::vector<webrtc::SdpAudioFormat>& formats =
+      decoder_factory_->GetSupportedFormats();
+
+  // Only generate CN payload types for these clockrates
+  std::map<int, bool, std::greater<int>> generate_cn = {{ 8000,  false },
+                                                        { 16000, false },
+                                                        { 32000, false }};
+
+  auto map_format = [&mapper, &out] (const webrtc::SdpAudioFormat& format) {
+    rtc::Optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
+    if (!opt_codec) {
+      LOG(LS_ERROR) << "Unable to assign payload type to format: " << format;
+      return false;
+    }
+
+    auto& codec = *opt_codec;
+    if (IsCodec(codec, kOpusCodecName)) {
+      // TODO(ossu): Set this specifically for Opus for now, until we have a
+      // better way of dealing with rtcp-fb parameters.
+      codec.AddFeedbackParam(
+          FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
+    }
+    out.push_back(codec);
+    return true;
+  };
+
+  for (const auto& format : formats) {
+    if (map_format(format)) {
+      // TODO(ossu): We should get more than just a format from the factory, so
+      // we can determine if a format should be used with CN or not. For now,
+      // generate a CN entry for each supported clock rate also used by a format
+      // supported by the factory.
+      auto cn = generate_cn.find(format.clockrate_hz);
+      if (cn != generate_cn.end() /* && format.allow_comfort_noise */) {
+        cn->second = true;
+      }
+    }
+  }
+
+  // Add CN codecs after "proper" audio codecs

[ossu, 2016/07/12 15:49:49]

I put these back after all the regular codecs, and in decreasing order of
clockrate, to mimic how they were previously ordered. It also makes sense, since
the list is supposed to be in priority order and in no case should a comfort
noise codec be higher priority than a "normal" one.

+  for (const auto& cn : generate_cn) {
+    if (cn.second) {
+      map_format({kCnCodecName, cn.first, 1});
+    }
+  }
+
+  // Add telephone-event codec last
+  map_format({kDtmfCodecName, 8000, 1});
+
+  return out;
+}
+
 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     : public AudioSource::Sink {
  public:
   WebRtcAudioSendStream(int ch,
                         webrtc::AudioTransport* voe_audio_transport,
                         uint32_t ssrc,
                         const std::string& c_name,
                         const SendCodecSpec& send_codec_spec,
                         const std::vector<webrtc::RtpExtension>& extensions,
                         webrtc::Call* call,
@@ skipping 1522 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE