b7e5054414
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example: * Change a few type declarations to better match how the majority of code uses those objects. * Eliminate "< 0" check for unsigned values. * Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar. * Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects. * Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t. * Similarly, add casts when passing a larger type to a function taking a smaller one. * Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar. * Use "false" instead of "0" for setting a bool. * Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t. BUG=none R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org TBR=andrew, asapersson, henrika Review URL: https://codereview.webrtc.org/1168753002 Cr-Commit-Position: refs/heads/master@{#9419}
115 lines
3.5 KiB
C++
115 lines
3.5 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 "webrtc/common_types.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/modules/utility/source/coder.h"
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namespace webrtc {
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AudioCoder::AudioCoder(uint32_t 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->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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}
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int32_t 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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int32_t 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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int32_t AudioCoder::Decode(AudioFrame& decodedAudio,
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uint32_t sampFreqHz,
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const int8_t* incomingPayload,
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size_t payloadLength)
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{
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if (payloadLength > 0)
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{
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const uint8_t payloadType = _receiveCodec.pltype;
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_decodeTimestamp += _receiveCodec.pacsize;
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if(_acm->IncomingPayload((const uint8_t*) 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((uint16_t)sampFreqHz, &decodedAudio);
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}
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int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio,
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uint16_t& sampFreqHz)
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{
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return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
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}
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int32_t AudioCoder::Encode(const AudioFrame& audio,
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int8_t* encodedData,
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size_t& 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 += static_cast<uint32_t>(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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encodedLengthInBytes = _encodedLengthInBytes;
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return 0;
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}
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int32_t AudioCoder::SendData(
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FrameType /* frameType */,
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uint8_t /* payloadType */,
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uint32_t /* timeStamp */,
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const uint8_t* payloadData,
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size_t payloadSize,
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const RTPFragmentationHeader* /* fragmentation*/)
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{
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memcpy(_encodedData,payloadData,sizeof(uint8_t) * 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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