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/opus/audio_decoder_opus.cc

@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2015 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/opus/audio_decoder_opus.h"
+
+#include <utility>
+
+#include "webrtc/rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+class OpusFrame : public AudioDecoder::EncodedAudioFrame {
+ public:
+  OpusFrame(AudioDecoderOpusImpl* decoder,
+            rtc::Buffer&& payload,
+            bool is_primary_payload)
+      : decoder_(decoder),
+        payload_(std::move(payload)),
+        is_primary_payload_(is_primary_payload) {}
+
+  size_t Duration() const override {
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    } else {
+      ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
+    }
+    return (ret < 0) ? 0 : static_cast<size_t>(ret);
+  }
+
+  rtc::Optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->Decode(
+          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    } else {
+      ret = decoder_->DecodeRedundant(
+          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});
+  }
+
+ private:
+  AudioDecoderOpusImpl* const decoder_;
+  const rtc::Buffer payload_;
+  const bool is_primary_payload_;
+};
+
+}  // namespace
+
+AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels)
+    : channels_(num_channels) {
+  RTC_DCHECK(num_channels == 1 || num_channels == 2);
+  WebRtcOpus_DecoderCreate(&dec_state_, channels_);
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderOpusImpl::~AudioDecoderOpusImpl() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+
+  if (PacketHasFec(payload.data(), payload.size())) {
+    const int duration =
+        PacketDurationRedundant(payload.data(), payload.size());
+    RTC_DCHECK_GE(duration, 0);
+    rtc::Buffer payload_copy(payload.data(), payload.size());
+    std::unique_ptr<EncodedAudioFrame> fec_frame(
+        new OpusFrame(this, std::move(payload_copy), false));
+    results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
+  }
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OpusFrame(this, std::move(payload), true));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded,
+                                                  size_t encoded_len,
+                                                  int sample_rate_hz,
+                                                  int16_t* decoded,
+                                                  SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderOpusImpl::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderOpusImpl::PacketDuration(const uint8_t* encoded,
+                                         size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded,
+                                                  size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len);
+}
+
+bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
+                                        size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+int AudioDecoderOpusImpl::SampleRateHz() const {
+  return 48000;
+}
+
+size_t AudioDecoderOpusImpl::Channels() const {
+  return channels_;
+}
+
+}  // namespace webrtc