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Run "git cl format --full" on a pair of files with ancient formatting

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Review-Url: https://codereview.webrtc.org/1946873003
Cr-Commit-Position: refs/heads/master@{#12625}
This commit is contained in:
kwiberg
2016-05-04 05:12:19 -07:00
committed by Commit bot
parent 053f917741
commit 73987c9932
2 changed files with 74 additions and 83 deletions
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37
webrtc/modules/utility/source/coder.cc
View File

@ -13,21 +13,19 @@
#include "webrtc/modules/utility/source/coder.h"
namespace webrtc {
AudioCoder::AudioCoder(uint32_t instanceID)
: _acm(AudioCodingModule::Create(instanceID)),
_receiveCodec(),
_encodeTimestamp(0),
_encodedData(NULL),
_encodedLengthInBytes(0),
_decodeTimestamp(0)
{
_decodeTimestamp(0) {
_acm->InitializeReceiver();
_acm->RegisterTransportCallback(this);
}
AudioCoder::~AudioCoder()
{
}
AudioCoder::~AudioCoder() {}
int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst) {
const bool success = codec_manager_.RegisterEncoder(codecInst) &&
@ -47,17 +45,12 @@ int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst) {
int32_t AudioCoder::Decode(AudioFrame& decodedAudio,
uint32_t sampFreqHz,
const int8_t* incomingPayload,
size_t payloadLength)
{
if (payloadLength > 0)
{
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)
{
if (_acm->IncomingPayload((const uint8_t*)incomingPayload, payloadLength,
payloadType, _decodeTimestamp) == -1) {
return -1;
}
}
@ -65,15 +58,13 @@ int32_t AudioCoder::Decode(AudioFrame& decodedAudio,
}
int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio,
uint16_t& sampFreqHz)
{
uint16_t& sampFreqHz) {
return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
}
int32_t AudioCoder::Encode(const AudioFrame& audio,
int8_t* encodedData,
size_t& encodedLengthInBytes)
{
size_t& 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;
@ -84,8 +75,7 @@ int32_t AudioCoder::Encode(const AudioFrame& audio,
// 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)
{
if (_acm->Add10MsData((AudioFrame&)audioFrame) == -1) {
return -1;
}
_encodedData = encodedData;
@ -93,16 +83,15 @@ int32_t AudioCoder::Encode(const AudioFrame& audio,
return 0;
}
int32_t AudioCoder::SendData(
FrameType /* frameType */,
int32_t AudioCoder::SendData(FrameType /* frameType */,
uint8_t /* payloadType */,
uint32_t /* timeStamp */,
const uint8_t* payloadData,
size_t payloadSize,
const RTPFragmentationHeader* /* fragmentation*/)
{
const RTPFragmentationHeader* /* fragmentation*/) {
memcpy(_encodedData, payloadData, sizeof(uint8_t) * payloadSize);
_encodedLengthInBytes = payloadSize;
return 0;
}
} // namespace webrtc

12
webrtc/modules/utility/source/coder.h
View File

@ -22,8 +22,7 @@
namespace webrtc {
class AudioFrame;
class AudioCoder : public AudioPacketizationCallback
{
class AudioCoder : public AudioPacketizationCallback {
public:
AudioCoder(uint32_t instanceID);
~AudioCoder();
@ -32,12 +31,15 @@ public:
int32_t SetDecodeCodec(const CodecInst& codecInst);
int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz,
const int8_t* incomingPayload, size_t payloadLength);
int32_t Decode(AudioFrame& decodedAudio,
uint32_t sampFreqHz,
const int8_t* incomingPayload,
size_t payloadLength);
int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz);
int32_t Encode(const AudioFrame& audio, int8_t* encodedData,
int32_t Encode(const AudioFrame& audio,
int8_t* encodedData,
size_t& encodedLengthInBytes);
protected: