ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
113 lines
3.4 KiB
C++
113 lines
3.4 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/modules/include/module_common_types.h"
|
|
#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);
|
|
}
|
|
|
|
AudioCoder::~AudioCoder()
|
|
{
|
|
}
|
|
|
|
int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst)
|
|
{
|
|
if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1)
|
|
{
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst)
|
|
{
|
|
if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1)
|
|
{
|
|
return -1;
|
|
}
|
|
memcpy(&_receiveCodec,&codecInst,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)
|
|
{
|
|
return -1;
|
|
}
|
|
}
|
|
return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio);
|
|
}
|
|
|
|
int32_t AudioCoder::PlayoutData(AudioFrame& 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_);
|
|
|
|
// 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;
|
|
}
|
|
} // namespace webrtc
|