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/audio_format_conversion.cc

@@ -0,0 +1,88 @@
+/*
+ *  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/audio_format_conversion.h"
+
+#include <string.h>
+
+#include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/safe_conversions.h"
+#include "webrtc/rtc_base/sanitizer.h"
+
+namespace webrtc {
+
+namespace {
+
+CodecInst MakeCodecInst(int payload_type,
+                        const char* name,
+                        int sample_rate,
+                        size_t num_channels) {
+  // Create a CodecInst with some fields set. The remaining fields are zeroed,
+  // but we tell MSan to consider them uninitialized.
+  CodecInst ci = {0};
+  rtc::MsanMarkUninitialized(rtc::MakeArrayView(&ci, 1));
+  ci.pltype = payload_type;
+  strncpy(ci.plname, name, sizeof(ci.plname));
+  ci.plname[sizeof(ci.plname) - 1] = '\0';
+  ci.plfreq = sample_rate;
+  ci.channels = num_channels;
+  return ci;
+}
+
+}  // namespace
+
+SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
+  if (STR_CASE_CMP(ci.plname, "g722") == 0) {
+    RTC_CHECK_EQ(16000, ci.plfreq);
+    RTC_CHECK(ci.channels == 1 || ci.channels == 2);
+    return {"g722", 8000, ci.channels};
+  } else if (STR_CASE_CMP(ci.plname, "opus") == 0) {
+    RTC_CHECK_EQ(48000, ci.plfreq);
+    RTC_CHECK(ci.channels == 1 || ci.channels == 2);
+    return ci.channels == 1
+               ? SdpAudioFormat("opus", 48000, 2)
+               : SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}});
+  } else {
+    return {ci.plname, ci.plfreq, ci.channels};
+  }
+}
+
+CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format) {
+  if (STR_CASE_CMP(audio_format.name.c_str(), "g722") == 0) {
+    RTC_CHECK_EQ(8000, audio_format.clockrate_hz);
+    RTC_CHECK(audio_format.num_channels == 1 || audio_format.num_channels == 2);
+    return MakeCodecInst(payload_type, "g722", 16000,
+                         audio_format.num_channels);
+  } else if (STR_CASE_CMP(audio_format.name.c_str(), "opus") == 0) {
+    RTC_CHECK_EQ(48000, audio_format.clockrate_hz);
+    RTC_CHECK_EQ(2, audio_format.num_channels);
+    const int num_channels = [&] {
+      auto stereo = audio_format.parameters.find("stereo");
+      if (stereo != audio_format.parameters.end()) {
+        if (stereo->second == "0") {
+          return 1;
+        } else if (stereo->second == "1") {
+          return 2;
+        } else {
+          RTC_CHECK(false);  // Bad stereo parameter.
+        }
+      }
+      return 1;  // Default to mono.
+    }();
+    return MakeCodecInst(payload_type, "opus", 48000, num_channels);
+  } else {
+    return MakeCodecInst(payload_type, audio_format.name.c_str(),
+                         audio_format.clockrate_hz, audio_format.num_channels);
+  }
+}
+
+}  // namespace webrtc