WIP: Move the creation of AudioCodecFactory into PeerConnectionFactory.

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/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.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"
@@ skipping 618 matching lines @@
   }
 
   return (highestNeeded);
 }
 
 int32_t Channel::CreateChannel(Channel*& channel,
                                int32_t channelId,
                                uint32_t instanceId,
                                RtcEventLog* const event_log,
                                const Config& config) {
+  return CreateChannel(channel, channelId, instanceId, event_log, config,
+                       CreateBuiltinAudioDecoderFactory());
+}
+
+int32_t Channel::CreateChannel(
+    Channel*& channel,
+    int32_t channelId,
+    uint32_t instanceId,
+    RtcEventLog* const event_log,
+    const Config& config,
+    std::shared_ptr<AudioDecoderFactory> decoder_factory) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                "Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
                instanceId);
 
-  channel = new Channel(channelId, instanceId, event_log, config);
+  channel = new Channel(channelId, instanceId, event_log, config,
+                        std::move(decoder_factory));
   if (channel == NULL) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                  "Channel::CreateChannel() unable to allocate memory for"
                  " channel");
     return -1;
   }
   return 0;
 }
 
 void Channel::PlayNotification(int32_t id, uint32_t durationMs) {
@@ skipping 39 matching lines @@
 
   _outputFileRecording = false;
   WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RecordFileEnded() => output file recorder module is"
                " shutdown");
 }
 
 Channel::Channel(int32_t channelId,
                  uint32_t instanceId,
                  RtcEventLog* const event_log,
-                 const Config& config)
+                 const Config& config,
+                 std::shared_ptr<AudioDecoderFactory> decoder_factory)
     : _instanceId(instanceId),
       _channelId(channelId),
       event_log_(event_log),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(
           new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
       rtp_receive_statistics_(
           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
       rtp_receiver_(
           RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
@@ skipping 54 matching lines @@
       _RxVadDetection(false),
       _rxAgcIsEnabled(false),
       _rxNsIsEnabled(false),
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
       associate_send_channel_(ChannelOwner(nullptr)),
       pacing_enabled_(config.Get<VoicePacing>().enabled),
       feedback_observer_proxy_(new TransportFeedbackProxy()),
       seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
-      rtp_packet_sender_proxy_(new RtpPacketSenderProxy()) {
+      rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
+      decoder_factory_(decoder_factory) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
   AudioCodingModule::Config acm_config;
   acm_config.id = VoEModuleId(instanceId, channelId);
   if (config.Get<NetEqCapacityConfig>().enabled) {
     // Clamping the buffer capacity at 20 packets. While going lower will
     // probably work, it makes little sense.
     acm_config.neteq_config.max_packets_in_buffer =
         std::max(20, config.Get<NetEqCapacityConfig>().capacity);
   }
   acm_config.neteq_config.enable_fast_accelerate =
       config.Get<NetEqFastAccelerate>().enabled;
+  acm_config.decoder_factory = std::move(decoder_factory);
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
   configuration.audio = true;
   configuration.outgoing_transport = this;
   configuration.receive_statistics = rtp_receive_statistics_.get();
   configuration.bandwidth_callback = rtcp_observer_.get();
   if (pacing_enabled_) {
@@ skipping 221 matching lines @@
 int32_t Channel::UpdateLocalTimeStamp() {
   _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
   return 0;
 }
 
 void Channel::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   rtc::CritScope cs(&_callbackCritSect);
   audio_sink_ = std::move(sink);
 }
 
+const std::shared_ptr<AudioDecoderFactory>&
+Channel::GetAudioDecoderFactory() const {
+  return decoder_factory_;
+}
 int32_t Channel::StartPlayout() {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::StartPlayout()");
   if (channel_state_.Get().playing) {
     return 0;
   }
 
   if (!_externalMixing) {
     // Add participant as candidates for mixing.
     if (_outputMixerPtr->SetMixabilityStatus(*this, true) != 0) {
@@ skipping 2481 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