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/codecs/legacy_encoded_audio_frame.cc

@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2016 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/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+namespace webrtc {
+
+LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
+                                                 rtc::Buffer&& payload)
+    : decoder_(decoder), payload_(std::move(payload)) {}
+
+LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default;
+
+size_t LegacyEncodedAudioFrame::Duration() const {
+  const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+  return (ret < 0) ? 0 : static_cast<size_t>(ret);
+}
+
+rtc::Optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
+LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
+  AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+  const int ret = decoder_->Decode(
+      payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+      decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+
+  if (ret < 0)
+    return rtc::Optional<DecodeResult>();
+
+  return rtc::Optional<DecodeResult>({static_cast<size_t>(ret), speech_type});
+}
+
+std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
+    AudioDecoder* decoder,
+    rtc::Buffer&& payload,
+    uint32_t timestamp,
+    size_t bytes_per_ms,
+    uint32_t timestamps_per_ms) {
+  RTC_DCHECK(payload.data());
+  std::vector<AudioDecoder::ParseResult> results;
+  size_t split_size_bytes = payload.size();
+
+  // Find a "chunk size" >= 20 ms and < 40 ms.
+  const size_t min_chunk_size = bytes_per_ms * 20;
+  if (min_chunk_size >= payload.size()) {
+    std::unique_ptr<LegacyEncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(decoder, std::move(payload)));
+    results.emplace_back(timestamp, 0, std::move(frame));
+  } else {
+    // Reduce the split size by half as long as |split_size_bytes| is at least
+    // twice the minimum chunk size (so that the resulting size is at least as
+    // large as the minimum chunk size).
+    while (split_size_bytes >= 2 * min_chunk_size) {
+      split_size_bytes /= 2;
+    }
+
+    const uint32_t timestamps_per_chunk = static_cast<uint32_t>(
+        split_size_bytes * timestamps_per_ms / bytes_per_ms);
+    size_t byte_offset;
+    uint32_t timestamp_offset;
+    for (byte_offset = 0, timestamp_offset = 0;
+         byte_offset < payload.size();
+         byte_offset += split_size_bytes,
+             timestamp_offset += timestamps_per_chunk) {
+      split_size_bytes =
+          std::min(split_size_bytes, payload.size() - byte_offset);
+      rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes);
+      std::unique_ptr<LegacyEncodedAudioFrame> frame(
+          new LegacyEncodedAudioFrame(decoder, std::move(new_payload)));
+      results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
+    }
+  }
+
+  return results;
+}
+
+}  // namespace webrtc