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/g711/audio_decoder_pcm.cc

@@ -0,0 +1,88 @@
+/*
+ *  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/g711/audio_decoder_pcm.h"
+
+#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
+
+namespace webrtc {
+
+void AudioDecoderPcmU::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmU::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmU::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+void AudioDecoderPcmA::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmA::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmA::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+}  // namespace webrtc