Run "git cl format --full" on a pair of files with ancient formatting
Review-Url: https://codereview.webrtc.org/1946873003 Cr-Commit-Position: refs/heads/master@{#12625}
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@ -13,21 +13,19 @@
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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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_decodeTimestamp(0) {
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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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AudioCoder::~AudioCoder() {}
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int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst) {
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const bool success = codec_manager_.RegisterEncoder(codecInst) &&
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@ -46,63 +44,54 @@ int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst) {
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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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const int8_t* incomingPayload,
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size_t payloadLength) {
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if (payloadLength > 0) {
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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, payloadLength,
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payloadType, _decodeTimestamp) == -1) {
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return -1;
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}
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return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio);
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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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uint16_t& sampFreqHz) {
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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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size_t& encodedLengthInBytes) {
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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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// 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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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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int32_t AudioCoder::SendData(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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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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