7a7a008031
Part of refactoring of ACM, and recent lint-warnings. This CL changes non-const references in the ACM API to pointers. BUG=issue1372 Committed: https://code.google.com/p/webrtc/source/detail?r=3543 Review URL: https://webrtc-codereview.appspot.com/1103012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3555 4adac7df-926f-26a2-2b94-8c16560cd09d
128 lines
3.6 KiB
C++
128 lines
3.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "coder.h"
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#include "common_types.h"
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#include "module_common_types.h"
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// OS independent case insensitive string comparison.
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#ifdef WIN32
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#define STR_CASE_CMP(x,y) ::_stricmp(x,y)
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#else
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#define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
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#endif
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namespace webrtc {
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AudioCoder::AudioCoder(WebRtc_UWord32 instanceID)
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: _acm(AudioCodingModule::Create(instanceID)),
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_receiveCodec(),
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_encodeTimestamp(0),
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_encodedData(NULL),
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_encodedLengthInBytes(0),
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_decodeTimestamp(0)
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{
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_acm->InitializeSender();
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_acm->InitializeReceiver();
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_acm->RegisterTransportCallback(this);
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}
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AudioCoder::~AudioCoder()
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{
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AudioCodingModule::Destroy(_acm);
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}
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WebRtc_Word32 AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
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ACMAMRPackingFormat amrFormat)
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{
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if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1)
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{
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return -1;
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}
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return 0;
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}
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WebRtc_Word32 AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
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ACMAMRPackingFormat amrFormat)
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{
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if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1)
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{
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return -1;
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}
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memcpy(&_receiveCodec,&codecInst,sizeof(CodecInst));
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return 0;
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}
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WebRtc_Word32 AudioCoder::Decode(AudioFrame& decodedAudio,
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WebRtc_UWord32 sampFreqHz,
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const WebRtc_Word8* incomingPayload,
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WebRtc_Word32 payloadLength)
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{
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if (payloadLength > 0)
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{
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const WebRtc_UWord8 payloadType = _receiveCodec.pltype;
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_decodeTimestamp += _receiveCodec.pacsize;
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if(_acm->IncomingPayload((const WebRtc_UWord8*) incomingPayload,
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payloadLength,
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payloadType,
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_decodeTimestamp) == -1)
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{
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return -1;
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}
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}
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return _acm->PlayoutData10Ms((WebRtc_UWord16)sampFreqHz, &decodedAudio);
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}
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WebRtc_Word32 AudioCoder::PlayoutData(AudioFrame& decodedAudio,
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WebRtc_UWord16& sampFreqHz)
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{
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return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
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}
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WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio,
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WebRtc_Word8* encodedData,
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WebRtc_UWord32& encodedLengthInBytes)
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{
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// Fake a timestamp in case audio doesn't contain a correct timestamp.
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// Make a local copy of the audio frame since audio is const
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AudioFrame audioFrame;
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audioFrame.CopyFrom(audio);
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audioFrame.timestamp_ = _encodeTimestamp;
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_encodeTimestamp += audioFrame.samples_per_channel_;
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// For any codec with a frame size that is longer than 10 ms the encoded
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// length in bytes should be zero until a a full frame has been encoded.
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_encodedLengthInBytes = 0;
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if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1)
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{
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return -1;
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}
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_encodedData = encodedData;
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if(_acm->Process() == -1)
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{
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return -1;
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}
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encodedLengthInBytes = _encodedLengthInBytes;
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return 0;
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}
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WebRtc_Word32 AudioCoder::SendData(
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FrameType /* frameType */,
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WebRtc_UWord8 /* payloadType */,
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WebRtc_UWord32 /* timeStamp */,
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const WebRtc_UWord8* payloadData,
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WebRtc_UWord16 payloadSize,
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const RTPFragmentationHeader* /* fragmentation*/)
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{
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memcpy(_encodedData,payloadData,sizeof(WebRtc_UWord8) * payloadSize);
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_encodedLengthInBytes = payloadSize;
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return 0;
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}
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} // namespace webrtc
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