ACM: Move NACK functionality inside NetEq

webrtc/modules/audio_coding/neteq/neteq_impl.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 15 matching lines @@
 #include "webrtc/modules/audio_coding/neteq/comfort_noise.h"
 #include "webrtc/modules/audio_coding/neteq/decision_logic.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/defines.h"
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
 #include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/merge.h"
+#include "webrtc/modules/audio_coding/neteq/nack.h"
 #include "webrtc/modules/audio_coding/neteq/normal.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
 #include "webrtc/modules/audio_coding/neteq/payload_splitter.h"
 #include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
 #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
@@ skipping 33 matching lines @@
       expand_factory_(expand_factory),
       accelerate_factory_(accelerate_factory),
       preemptive_expand_factory_(preemptive_expand_factory),
       last_mode_(kModeNormal),
       decoded_buffer_length_(kMaxFrameSize),
       decoded_buffer_(new int16_t[decoded_buffer_length_]),
       playout_timestamp_(0),
       new_codec_(false),
       timestamp_(0),
       reset_decoder_(false),
-      current_rtp_payload_type_(0xFF),  // Invalid RTP payload type.
+      current_rtp_payload_type_(0xFF),      // Invalid RTP payload type.
       current_cng_rtp_payload_type_(0xFF),  // Invalid RTP payload type.
       ssrc_(0),
       first_packet_(true),
       error_code_(0),
       decoder_error_code_(0),
       background_noise_mode_(config.background_noise_mode),
       playout_mode_(config.playout_mode),
       enable_fast_accelerate_(config.enable_fast_accelerate),
-      decoded_packet_sequence_number_(-1),
-      decoded_packet_timestamp_(0) {
+      nack_enabled_(false) {
   LOG(LS_INFO) << "NetEq config: " << config.ToString();
   int fs = config.sample_rate_hz;
   if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
     LOG(LS_ERROR) << "Sample rate " << fs << " Hz not supported. " <<
         "Changing to 8000 Hz.";
     fs = 8000;
   }
   fs_hz_ = fs;
   fs_mult_ = fs / 8000;
   output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
@@ skipping 288 matching lines @@
   // Set to wait for new codec.
   first_packet_ = true;
 }
 
 void NetEqImpl::PacketBufferStatistics(int* current_num_packets,
                                        int* max_num_packets) const {
   CriticalSectionScoped lock(crit_sect_.get());
   packet_buffer_->BufferStat(current_num_packets, max_num_packets);
 }
 
-int NetEqImpl::DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const {
+void NetEqImpl::EnableNack(size_t max_nack_list_size) {
+  RTC_CHECK_GT(max_nack_list_size, 0u);
+  // Ugly hack to get around the problem of passing static consts by reference.
+  const size_t kNackListSizeLimitLocal = Nack::kNackListSizeLimit;
+  RTC_CHECK_LE(max_nack_list_size, kNackListSizeLimitLocal);
+
   CriticalSectionScoped lock(crit_sect_.get());
-  if (decoded_packet_sequence_number_ < 0)
-    return -1;
-  *sequence_number = decoded_packet_sequence_number_;
-  *timestamp = decoded_packet_timestamp_;
-  return 0;
+  if (!nack_enabled_) {
+    const int kNackThresholdPackets = 2;
+    nack_.reset(Nack::Create(kNackThresholdPackets));
+    nack_enabled_ = true;
+    nack_->UpdateSampleRate(fs_hz_);
+  }
+  RTC_CHECK_EQ(nack_->SetMaxNackListSize(max_nack_list_size), 0);

[minyue-webrtc, 2015/10/27 14:35:08]

why not make SetMaxNackListSize() a void and move 409 to 412 in to it

[hlundin-webrtc, 2015/10/28 15:03:36]

Good idea. Done.

+}
+
+void NetEqImpl::DisableNack() {
+  CriticalSectionScoped lock(crit_sect_.get());
+  nack_.reset();
+  nack_enabled_ = false;
+}
+
+std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const {
+  RTC_CHECK_GE(round_trip_time_ms, 0);

[minyue-webrtc, 2015/10/27 14:35:08]

is this necessary to crash?

[hlundin-webrtc, 2015/10/28 15:03:36]

No, you are right. Changed to DCHECK. But I also moved it into
Nack::GetNackList.

+  CriticalSectionScoped lock(crit_sect_.get());
+  if (!nack_enabled_) {
+    return std::vector<uint16_t>();
+  }
+  RTC_DCHECK(nack_.get());
+  return nack_->GetNackList(round_trip_time_ms);
 }
 
 const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
   CriticalSectionScoped lock(crit_sect_.get());
   return sync_buffer_.get();
 }
 
 // Methods below this line are private.
 
 int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
@@ skipping 179 matching lines @@
         decoder_database_->GetDecoder(main_header.payloadType);
     assert(decoder);  // Should always get a valid object, since we have
                       // already checked that the payload types are known.
     decoder->IncomingPacket(packet_list.front()->payload,
                             packet_list.front()->payload_length,
                             packet_list.front()->header.sequenceNumber,
                             packet_list.front()->header.timestamp,
                             receive_timestamp);
   }
 
+  if (nack_enabled_) {
+    RTC_DCHECK(nack_);

[minyue-webrtc, 2015/10/27 14:35:08]

nack_.get()?

[hlundin-webrtc, 2015/10/28 15:03:36]

Not needed. This expands "in the obvious way" through some clever tricks in
scoped_ptr.

[minyue-webrtc, 2015/10/29 10:36:10]

Ack, but we use get() elsewhere a lot, see line 436. I let you decide.

[hlundin-webrtc, 2015/10/29 10:38:18]

I know, but I'm trying to do better now. :)

[kwiberg-webrtc, 2015/10/29 10:46:01]

+1

+    if (update_sample_rate_and_channels) {
+      nack_->Reset();
+    }
+    nack_->UpdateLastReceivedPacket(packet_list.front()->header.sequenceNumber,
+                                    packet_list.front()->header.timestamp);
+  }
+
   // Insert packets in buffer.
   const size_t buffer_length_before_insert =
       packet_buffer_->NumPacketsInBuffer();
   ret = packet_buffer_->InsertPacketList(
       &packet_list,
       *decoder_database_,
       &current_rtp_payload_type_,
       &current_cng_rtp_payload_type_);
   if (ret == PacketBuffer::kFlushed) {
     // Reset DSP timestamp etc. if packet buffer flushed.
@@ skipping 27 matching lines @@
     // payload type of the hypothetically new send codec is not known.
     const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader();
     assert(rtp_header);
     int payload_type = rtp_header->payloadType;
     AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type);
     assert(decoder);  // Payloads are already checked to be valid.
     const DecoderDatabase::DecoderInfo* decoder_info =
         decoder_database_->GetDecoderInfo(payload_type);
     assert(decoder_info);
     if (decoder_info->fs_hz != fs_hz_ ||
-        decoder->Channels() != algorithm_buffer_->Channels())
+        decoder->Channels() != algorithm_buffer_->Channels()) {
       SetSampleRateAndChannels(decoder_info->fs_hz, decoder->Channels());
+    }
+    if (nack_enabled_) {
+      RTC_DCHECK(nack_);
+      // Update the sample rate even if the rate is not new, because of Reset().
+      nack_->UpdateSampleRate(fs_hz_);
+    }
   }
 
   // TODO(hlundin): Move this code to DelayManager class.
   const DecoderDatabase::DecoderInfo* dec_info =
           decoder_database_->GetDecoderInfo(main_header.payloadType);
   assert(dec_info);  // Already checked that the payload type is known.
   delay_manager_->LastDecoderType(dec_info->codec_type);
   if (delay_manager_->last_pack_cng_or_dtmf() == 0) {
     // Calculate the total speech length carried in each packet.
     const size_t buffer_length_after_insert =
@@ skipping 1183 matching lines @@
       return -1;
     }
     stats_.PacketsDiscarded(discard_count);
     // Store waiting time in ms; packets->waiting_time is in "output blocks".
     stats_.StoreWaitingTime(packet->waiting_time * kOutputSizeMs);
     assert(packet->payload_length > 0);
     packet_list->push_back(packet);  // Store packet in list.
 
     if (first_packet) {
       first_packet = false;
-      decoded_packet_sequence_number_ = prev_sequence_number =
-          packet->header.sequenceNumber;
-      decoded_packet_timestamp_ = prev_timestamp = packet->header.timestamp;
+      if (nack_enabled_) {
+        RTC_DCHECK(nack_);
+        // TODO(henrik.lundin): Should we update this for all decoded packets?
+        nack_->UpdateLastDecodedPacket(packet->header.sequenceNumber,
+                                       packet->header.timestamp);
+      }
+      prev_sequence_number = packet->header.sequenceNumber;
+      prev_timestamp = packet->header.timestamp;
       prev_payload_type = packet->header.payloadType;
     }
 
     // Store number of extracted samples.
     int packet_duration = 0;
     AudioDecoder* decoder = decoder_database_->GetDecoder(
         packet->header.payloadType);
     if (decoder) {
       if (packet->sync_packet) {
         packet_duration = rtc::checked_cast<int>(decoder_frame_length_);
@@ skipping 147 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc