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}
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index 1476e02..a376cc7 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -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)
-{
- _acm->InitializeReceiver();
- _acm->RegisterTransportCallback(this);
+ _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) &&
@@ -46,63 +44,54 @@
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)
- {
- return -1;
- }
+ 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) {
+ return -1;
}
- return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio);
+ }
+ return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio);
}
int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio,
- uint16_t& sampFreqHz)
-{
- return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
+ uint16_t& sampFreqHz) {
+ return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
}
int32_t AudioCoder::Encode(const AudioFrame& audio,
int8_t* encodedData,
- 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;
- audioFrame.CopyFrom(audio);
- audioFrame.timestamp_ = _encodeTimestamp;
- _encodeTimestamp += static_cast<uint32_t>(audioFrame.samples_per_channel_);
+ 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;
+ audioFrame.CopyFrom(audio);
+ audioFrame.timestamp_ = _encodeTimestamp;
+ _encodeTimestamp += static_cast<uint32_t>(audioFrame.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)
- {
- return -1;
- }
- _encodedData = encodedData;
- encodedLengthInBytes = _encodedLengthInBytes;
- return 0;
+ // 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) {
+ return -1;
+ }
+ _encodedData = encodedData;
+ encodedLengthInBytes = _encodedLengthInBytes;
+ return 0;
}
-int32_t AudioCoder::SendData(
- FrameType /* frameType */,
- uint8_t /* payloadType */,
- uint32_t /* timeStamp */,
- const uint8_t* payloadData,
- size_t payloadSize,
- const RTPFragmentationHeader* /* fragmentation*/)
-{
- memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
- _encodedLengthInBytes = payloadSize;
- return 0;
+int32_t AudioCoder::SendData(FrameType /* frameType */,
+ uint8_t /* payloadType */,
+ uint32_t /* timeStamp */,
+ const uint8_t* payloadData,
+ size_t payloadSize,
+ const RTPFragmentationHeader* /* fragmentation*/) {
+ memcpy(_encodedData, payloadData, sizeof(uint8_t) * payloadSize);
+ _encodedLengthInBytes = payloadSize;
+ return 0;
}
+
} // namespace webrtc