Revert of Move FilePlayer and FileRecorder to Voice Engine

webrtc/modules/utility/source/coder.cc

Index: webrtc/modules/utility/source/coder.cc
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
new file mode 100644
index 0000000000000000000000000000000000000000..f2ae43eb108a939f261ae17646a05c26fd307643
--- /dev/null
+++ b/webrtc/modules/utility/source/coder.cc
@@ -0,0 +1,116 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/source/coder.h"
+
+namespace webrtc {
+namespace {
+AudioCodingModule::Config GetAcmConfig(uint32_t id) {
+  AudioCodingModule::Config config;
+  // This class does not handle muted output.
+  config.neteq_config.enable_muted_state = false;
+  config.id = id;
+  config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+  return config;
+}
+}  // namespace
+
+AudioCoder::AudioCoder(uint32_t instance_id)
+    : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))),
+      receive_codec_(),
+      encode_timestamp_(0),
+      encoded_data_(nullptr),
+      encoded_length_in_bytes_(0),
+      decode_timestamp_(0) {
+  acm_->InitializeReceiver();
+  acm_->RegisterTransportCallback(this);
+}
+
+AudioCoder::~AudioCoder() {}
+
+int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) {
+  const bool success = codec_manager_.RegisterEncoder(codec_inst) &&
+                       codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get());
+  return success ? 0 : -1;
+}
+
+int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
+  if (acm_->RegisterReceiveCodec(codec_inst, [&] {
+        return rent_a_codec_.RentIsacDecoder(codec_inst.plfreq);
+      }) == -1) {
+    return -1;
+  }
+  memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
+  return 0;
+}
+
+int32_t AudioCoder::Decode(AudioFrame& decoded_audio,
+                           uint32_t samp_freq_hz,
+                           const int8_t* incoming_payload,
+                           size_t payload_length) {
+  if (payload_length > 0) {
+    const uint8_t payload_type = receive_codec_.pltype;
+    decode_timestamp_ += receive_codec_.pacsize;
+    if (acm_->IncomingPayload((const uint8_t*)incoming_payload, payload_length,
+                              payload_type, decode_timestamp_) == -1) {
+      return -1;
+    }
+  }
+  bool muted;
+  int32_t ret =
+      acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio, &muted);
+  RTC_DCHECK(!muted);
+  return ret;
+}
+
+int32_t AudioCoder::PlayoutData(AudioFrame& decoded_audio,
+                                uint16_t& samp_freq_hz) {
+  bool muted;
+  int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio, &muted);
+  RTC_DCHECK(!muted);
+  return ret;
+}
+
+int32_t AudioCoder::Encode(const AudioFrame& audio,
+                           int8_t* encoded_data,
+                           size_t& encoded_length_in_bytes) {
+  // Fake a timestamp in case audio doesn't contain a correct timestamp.
+  // Make a local copy of the audio frame since audio is const
+  AudioFrame audio_frame;
+  audio_frame.CopyFrom(audio);
+  audio_frame.timestamp_ = encode_timestamp_;
+  encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_);
+
+  // For any codec with a frame size that is longer than 10 ms the encoded
+  // length in bytes should be zero until a a full frame has been encoded.
+  encoded_length_in_bytes_ = 0;
+  if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
+    return -1;
+  }
+  encoded_data_ = encoded_data;
+  encoded_length_in_bytes = encoded_length_in_bytes_;
+  return 0;
+}
+
+int32_t AudioCoder::SendData(FrameType /* frame_type */,
+                             uint8_t /* payload_type */,
+                             uint32_t /* time_stamp */,
+                             const uint8_t* payload_data,
+                             size_t payload_size,
+                             const RTPFragmentationHeader* /* fragmentation*/) {
+  memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
+  encoded_length_in_bytes_ = payload_size;
+  return 0;
+}
+
+}  // namespace webrtc