| Index: webrtc/modules/utility/source/coder.cc
|
| diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
|
| index a376cc79273286420fd3898faddaefa21655f753..9e43ca8df7629039cc24c45c4321b50c4fc18d62 100644
|
| --- a/webrtc/modules/utility/source/coder.cc
|
| +++ b/webrtc/modules/utility/source/coder.cc
|
| @@ -14,83 +14,83 @@
|
|
|
| namespace webrtc {
|
|
|
| -AudioCoder::AudioCoder(uint32_t instanceID)
|
| - : _acm(AudioCodingModule::Create(instanceID)),
|
| - _receiveCodec(),
|
| - _encodeTimestamp(0),
|
| - _encodedData(NULL),
|
| - _encodedLengthInBytes(0),
|
| - _decodeTimestamp(0) {
|
| - _acm->InitializeReceiver();
|
| - _acm->RegisterTransportCallback(this);
|
| +AudioCoder::AudioCoder(uint32_t instance_id)
|
| + : acm_(AudioCodingModule::Create(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& codecInst) {
|
| - const bool success = codec_manager_.RegisterEncoder(codecInst) &&
|
| - codec_manager_.MakeEncoder(&rent_a_codec_, _acm.get());
|
| +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& codecInst) {
|
| - if (_acm->RegisterReceiveCodec(
|
| - codecInst, [&] { return rent_a_codec_.RentIsacDecoder(); }) == -1) {
|
| +int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
|
| + if (acm_->RegisterReceiveCodec(
|
| + codec_inst, [&] { return rent_a_codec_.RentIsacDecoder(); }) == -1) {
|
| return -1;
|
| }
|
| - memcpy(&_receiveCodec, &codecInst, sizeof(CodecInst));
|
| + memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
|
| return 0;
|
| }
|
|
|
| -int32_t AudioCoder::Decode(AudioFrame& decodedAudio,
|
| - uint32_t sampFreqHz,
|
| - const int8_t* incomingPayload,
|
| - size_t payloadLength) {
|
| - if (payloadLength > 0) {
|
| - const uint8_t payloadType = _receiveCodec.pltype;
|
| - _decodeTimestamp += _receiveCodec.pacsize;
|
| - if (_acm->IncomingPayload((const uint8_t*)incomingPayload, payloadLength,
|
| - payloadType, _decodeTimestamp) == -1) {
|
| +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;
|
| }
|
| }
|
| - return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio);
|
| + return acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio);
|
| }
|
|
|
| -int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio,
|
| - uint16_t& sampFreqHz) {
|
| - return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
|
| +int32_t AudioCoder::PlayoutData(AudioFrame& decoded_audio,
|
| + uint16_t& samp_freq_hz) {
|
| + return acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio);
|
| }
|
|
|
| int32_t AudioCoder::Encode(const AudioFrame& audio,
|
| - int8_t* encodedData,
|
| - size_t& encodedLengthInBytes) {
|
| + 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 audioFrame;
|
| - audioFrame.CopyFrom(audio);
|
| - audioFrame.timestamp_ = _encodeTimestamp;
|
| - _encodeTimestamp += static_cast<uint32_t>(audioFrame.samples_per_channel_);
|
| + 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.
|
| - _encodedLengthInBytes = 0;
|
| - if (_acm->Add10MsData((AudioFrame&)audioFrame) == -1) {
|
| + encoded_length_in_bytes_ = 0;
|
| + if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
|
| return -1;
|
| }
|
| - _encodedData = encodedData;
|
| - encodedLengthInBytes = _encodedLengthInBytes;
|
| + encoded_data_ = encoded_data;
|
| + encoded_length_in_bytes = encoded_length_in_bytes_;
|
| return 0;
|
| }
|
|
|
| -int32_t AudioCoder::SendData(FrameType /* frameType */,
|
| - uint8_t /* payloadType */,
|
| - uint32_t /* timeStamp */,
|
| - const uint8_t* payloadData,
|
| - size_t payloadSize,
|
| +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(_encodedData, payloadData, sizeof(uint8_t) * payloadSize);
|
| - _encodedLengthInBytes = payloadSize;
|
| + memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
|
| + encoded_length_in_bytes_ = payload_size;
|
| return 0;
|
| }
|
|
|
|
|