git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/utility/source/coder.cc b/modules/utility/source/coder.cc
new file mode 100644
index 0000000..b858da1
--- /dev/null
+++ b/modules/utility/source/coder.cc
@@ -0,0 +1,128 @@
+/*
+ * Copyright (c) 2011 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 "coder.h"
+#include "common_types.h"
+#include "module_common_types.h"
+
+// OS independent case insensitive string comparison.
+#ifdef WIN32
+ #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
+#else
+ #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
+#endif
+
+namespace webrtc {
+AudioCoder::AudioCoder(WebRtc_UWord32 instanceID)
+ : _instanceID(instanceID),
+ _acm(AudioCodingModule::Create(instanceID)),
+ _receiveCodec(),
+ _encodeTimestamp(0),
+ _encodedData(NULL),
+ _encodedLengthInBytes(0),
+ _decodeTimestamp(0)
+{
+ _acm->InitializeSender();
+ _acm->InitializeReceiver();
+ _acm->RegisterTransportCallback(this);
+}
+
+AudioCoder::~AudioCoder()
+{
+ AudioCodingModule::Destroy(_acm);
+}
+
+WebRtc_Word32 AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
+ ACMAMRPackingFormat amrFormat)
+{
+ if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1)
+ {
+ return -1;
+ }
+ return 0;
+}
+
+WebRtc_Word32 AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
+ ACMAMRPackingFormat amrFormat)
+{
+ if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1)
+ {
+ return -1;
+ }
+ memcpy(&_receiveCodec,&codecInst,sizeof(CodecInst));
+ return 0;
+}
+
+WebRtc_Word32 AudioCoder::Decode(AudioFrame& decodedAudio,
+ WebRtc_UWord32 sampFreqHz,
+ const WebRtc_Word8* incomingPayload,
+ WebRtc_Word32 payloadLength)
+{
+ if (payloadLength > 0)
+ {
+ const WebRtc_UWord8 payloadType = _receiveCodec.pltype;
+ _decodeTimestamp += _receiveCodec.pacsize;
+ if(_acm->IncomingPayload(incomingPayload,
+ payloadLength,
+ payloadType,
+ _decodeTimestamp) == -1)
+ {
+ return -1;
+ }
+ }
+ return _acm->PlayoutData10Ms((WebRtc_UWord16)sampFreqHz,
+ (AudioFrame&)decodedAudio);
+}
+
+WebRtc_Word32 AudioCoder::PlayoutData(AudioFrame& decodedAudio,
+ WebRtc_UWord16& sampFreqHz)
+{
+ return _acm->PlayoutData10Ms(sampFreqHz, (AudioFrame&)decodedAudio);
+}
+
+WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio,
+ WebRtc_Word8* encodedData,
+ WebRtc_UWord32& encodedLengthInBytes)
+{
+ // 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 audioFrame = audio;
+ audioFrame._timeStamp = _encodeTimestamp;
+ _encodeTimestamp += audioFrame._payloadDataLengthInSamples;
+
+ // 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.
+ _encodedLengthInBytes = 0;
+ if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1)
+ {
+ return -1;
+ }
+ _encodedData = encodedData;
+ if(_acm->Process() == -1)
+ {
+ return -1;
+ }
+ encodedLengthInBytes = _encodedLengthInBytes;
+ return 0;
+}
+
+WebRtc_Word32 AudioCoder::SendData(
+ FrameType /* frameType */,
+ WebRtc_UWord8 /* payloadType */,
+ WebRtc_UWord32 /* timeStamp */,
+ const WebRtc_UWord8* payloadData,
+ WebRtc_UWord16 payloadSize,
+ const RTPFragmentationHeader* /* fragmentation*/)
+{
+ memcpy(_encodedData,payloadData,sizeof(WebRtc_UWord8) * payloadSize);
+ _encodedLengthInBytes = payloadSize;
+ return 0;
+}
+} // namespace webrtc