Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/neteq/packet_buffer.cc

@@ -0,0 +1,294 @@
+/*
+ *  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.
+ */
+
+// This is the implementation of the PacketBuffer class. It is mostly based on
+// an STL list. The list is kept sorted at all times so that the next packet to
+// decode is at the beginning of the list.
+
+#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
+
+#include <algorithm>  // find_if()
+
+#include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
+#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
+#include "webrtc/rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+// Predicate used when inserting packets in the buffer list.
+// Operator() returns true when |packet| goes before |new_packet|.
+class NewTimestampIsLarger {
+ public:
+  explicit NewTimestampIsLarger(const Packet& new_packet)
+      : new_packet_(new_packet) {
+  }
+  bool operator()(const Packet& packet) {
+    return (new_packet_ >= packet);
+  }
+
+ private:
+  const Packet& new_packet_;
+};
+
+// Returns true if both payload types are known to the decoder database, and
+// have the same sample rate.
+bool EqualSampleRates(uint8_t pt1,
+                      uint8_t pt2,
+                      const DecoderDatabase& decoder_database) {
+  auto* di1 = decoder_database.GetDecoderInfo(pt1);
+  auto* di2 = decoder_database.GetDecoderInfo(pt2);
+  return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz();
+}
+
+void LogPacketDiscarded(int codec_level, StatisticsCalculator* stats) {
+  RTC_CHECK(stats);
+  if (codec_level > 0) {
+    stats->SecondaryPacketsDiscarded(1);
+  } else {
+    stats->PacketsDiscarded(1);
+  }
+}
+
+}  // namespace
+
+PacketBuffer::PacketBuffer(size_t max_number_of_packets,
+                           const TickTimer* tick_timer)
+    : max_number_of_packets_(max_number_of_packets), tick_timer_(tick_timer) {}
+
+// Destructor. All packets in the buffer will be destroyed.
+PacketBuffer::~PacketBuffer() {
+  Flush();
+}
+
+// Flush the buffer. All packets in the buffer will be destroyed.
+void PacketBuffer::Flush() {
+  buffer_.clear();
+}
+
+bool PacketBuffer::Empty() const {
+  return buffer_.empty();
+}
+
+int PacketBuffer::InsertPacket(Packet&& packet, StatisticsCalculator* stats) {
+  if (packet.empty()) {
+    LOG(LS_WARNING) << "InsertPacket invalid packet";
+    return kInvalidPacket;
+  }
+
+  RTC_DCHECK_GE(packet.priority.codec_level, 0);
+  RTC_DCHECK_GE(packet.priority.red_level, 0);
+
+  int return_val = kOK;
+
+  packet.waiting_time = tick_timer_->GetNewStopwatch();
+
+  if (buffer_.size() >= max_number_of_packets_) {
+    // Buffer is full. Flush it.
+    Flush();
+    LOG(LS_WARNING) << "Packet buffer flushed";
+    return_val = kFlushed;
+  }
+
+  // Get an iterator pointing to the place in the buffer where the new packet
+  // should be inserted. The list is searched from the back, since the most
+  // likely case is that the new packet should be near the end of the list.
+  PacketList::reverse_iterator rit = std::find_if(
+      buffer_.rbegin(), buffer_.rend(),
+      NewTimestampIsLarger(packet));
+
+  // The new packet is to be inserted to the right of |rit|. If it has the same
+  // timestamp as |rit|, which has a higher priority, do not insert the new
+  // packet to list.
+  if (rit != buffer_.rend() && packet.timestamp == rit->timestamp) {
+    LogPacketDiscarded(packet.priority.codec_level, stats);
+    return return_val;
+  }
+
+  // The new packet is to be inserted to the left of |it|. If it has the same
+  // timestamp as |it|, which has a lower priority, replace |it| with the new
+  // packet.
+  PacketList::iterator it = rit.base();
+  if (it != buffer_.end() && packet.timestamp == it->timestamp) {
+    LogPacketDiscarded(packet.priority.codec_level, stats);
+    it = buffer_.erase(it);
+  }
+  buffer_.insert(it, std::move(packet));  // Insert the packet at that position.
+
+  return return_val;
+}
+
+int PacketBuffer::InsertPacketList(
+    PacketList* packet_list,
+    const DecoderDatabase& decoder_database,
+    rtc::Optional<uint8_t>* current_rtp_payload_type,
+    rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
+    StatisticsCalculator* stats) {
+  RTC_DCHECK(stats);
+  bool flushed = false;
+  for (auto& packet : *packet_list) {
+    if (decoder_database.IsComfortNoise(packet.payload_type)) {
+      if (*current_cng_rtp_payload_type &&
+          **current_cng_rtp_payload_type != packet.payload_type) {
+        // New CNG payload type implies new codec type.
+        *current_rtp_payload_type = rtc::Optional<uint8_t>();
+        Flush();
+        flushed = true;
+      }
+      *current_cng_rtp_payload_type =
+          rtc::Optional<uint8_t>(packet.payload_type);
+    } else if (!decoder_database.IsDtmf(packet.payload_type)) {
+      // This must be speech.
+      if ((*current_rtp_payload_type &&
+           **current_rtp_payload_type != packet.payload_type) ||
+          (*current_cng_rtp_payload_type &&
+           !EqualSampleRates(packet.payload_type,
+                             **current_cng_rtp_payload_type,
+                             decoder_database))) {
+        *current_cng_rtp_payload_type = rtc::Optional<uint8_t>();
+        Flush();
+        flushed = true;
+      }
+      *current_rtp_payload_type = rtc::Optional<uint8_t>(packet.payload_type);
+    }
+    int return_val = InsertPacket(std::move(packet), stats);
+    if (return_val == kFlushed) {
+      // The buffer flushed, but this is not an error. We can still continue.
+      flushed = true;
+    } else if (return_val != kOK) {
+      // An error occurred. Delete remaining packets in list and return.
+      packet_list->clear();
+      return return_val;
+    }
+  }
+  packet_list->clear();
+  return flushed ? kFlushed : kOK;
+}
+
+int PacketBuffer::NextTimestamp(uint32_t* next_timestamp) const {
+  if (Empty()) {
+    return kBufferEmpty;
+  }
+  if (!next_timestamp) {
+    return kInvalidPointer;
+  }
+  *next_timestamp = buffer_.front().timestamp;
+  return kOK;
+}
+
+int PacketBuffer::NextHigherTimestamp(uint32_t timestamp,
+                                      uint32_t* next_timestamp) const {
+  if (Empty()) {
+    return kBufferEmpty;
+  }
+  if (!next_timestamp) {
+    return kInvalidPointer;
+  }
+  PacketList::const_iterator it;
+  for (it = buffer_.begin(); it != buffer_.end(); ++it) {
+    if (it->timestamp >= timestamp) {
+      // Found a packet matching the search.
+      *next_timestamp = it->timestamp;
+      return kOK;
+    }
+  }
+  return kNotFound;
+}
+
+const Packet* PacketBuffer::PeekNextPacket() const {
+  return buffer_.empty() ? nullptr : &buffer_.front();
+}
+
+rtc::Optional<Packet> PacketBuffer::GetNextPacket() {
+  if (Empty()) {
+    // Buffer is empty.
+    return rtc::Optional<Packet>();
+  }
+
+  rtc::Optional<Packet> packet(std::move(buffer_.front()));
+  // Assert that the packet sanity checks in InsertPacket method works.
+  RTC_DCHECK(!packet->empty());
+  buffer_.pop_front();
+
+  return packet;
+}
+
+int PacketBuffer::DiscardNextPacket(StatisticsCalculator* stats) {
+  if (Empty()) {
+    return kBufferEmpty;
+  }
+  // Assert that the packet sanity checks in InsertPacket method works.
+  const Packet& packet = buffer_.front();
+  RTC_DCHECK(!packet.empty());
+  LogPacketDiscarded(packet.priority.codec_level, stats);
+  buffer_.pop_front();
+  return kOK;
+}
+
+void PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit,
+                                     uint32_t horizon_samples,
+                                     StatisticsCalculator* stats) {
+  buffer_.remove_if([timestamp_limit, horizon_samples, stats](const Packet& p) {
+    if (timestamp_limit == p.timestamp ||
+        !IsObsoleteTimestamp(p.timestamp, timestamp_limit, horizon_samples)) {
+      return false;
+    }
+    LogPacketDiscarded(p.priority.codec_level, stats);
+    return true;
+  });
+}
+
+void PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit,
+                                        StatisticsCalculator* stats) {
+  DiscardOldPackets(timestamp_limit, 0, stats);
+}
+
+void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type,
+                                                 StatisticsCalculator* stats) {
+  buffer_.remove_if([payload_type, stats](const Packet& p) {
+    if (p.payload_type != payload_type) {
+      return false;
+    }
+    LogPacketDiscarded(p.priority.codec_level, stats);
+    return true;
+  });
+}
+
+size_t PacketBuffer::NumPacketsInBuffer() const {
+  return buffer_.size();
+}
+
+size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
+  size_t num_samples = 0;
+  size_t last_duration = last_decoded_length;
+  for (const Packet& packet : buffer_) {
+    if (packet.frame) {
+      // TODO(hlundin): Verify that it's fine to count all packets and remove
+      // this check.
+      if (packet.priority != Packet::Priority(0, 0)) {
+        continue;
+      }
+      size_t duration = packet.frame->Duration();
+      if (duration > 0) {
+        last_duration = duration;  // Save the most up-to-date (valid) duration.
+      }
+    }
+    num_samples += last_duration;
+  }
+  return num_samples;
+}
+
+void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
+  *num_packets = static_cast<int>(buffer_.size());
+  *max_num_packets = static_cast<int>(max_number_of_packets_);
+}
+
+}  // namespace webrtc