Files
rtphone/src/engine/media/MT_SingleAudioStream.cpp
T

62 lines
2.0 KiB
C++

/* Copyright(C) 2007-2018 VoIP objects (voipobjects.com)
* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "MT_SingleAudioStream.h"
#include "MT_CodecList.h"
//#include "resip/stack/SdpContents.hxx"
#include "../engine/helper/HL_Log.h"
#define LOG_SUBSYSTEM "media"
using namespace MT;
SingleAudioStream::SingleAudioStream(const CodecList::Settings& codecSettings, Statistics& stat)
:mReceiver(codecSettings, stat),
mDtmfReceiver(stat)
{}
SingleAudioStream::~SingleAudioStream()
{}
void SingleAudioStream::process(const std::shared_ptr<jrtplib::RTPPacket>& packet)
{
ICELogMedia(<< "Processing incoming RTP/RTCP packet");
if (packet->GetPayloadType() == mReceiver.getCodecSettings().mTelephoneEvent)
mDtmfReceiver.add(packet);
else
mReceiver.add(packet);
}
void SingleAudioStream::copyPcmTo(Audio::DataWindow& output, int needed)
{
// Packet by packet
while (output.filled() < needed)
{
// Number of bytes to fill on this step
auto requested = needed - output.filled();
// requested is in bytes: 16-bit samples at AUDIO_SAMPLERATE/AUDIO_CHANNELS
constexpr int bytesPerMs = AUDIO_SAMPLERATE / 1000 * sizeof(int16_t) * AUDIO_CHANNELS;
auto options = AudioReceiver::DecodeOptions{
.mRealtimeProcessing = true,
.mResampleToMainRate = true,
.mSkipDecode = false,
.mElapsed = std::chrono::milliseconds(requested / bytesPerMs)
};
// Try to get the data from receiver / decoder
if (options.mElapsed != 0ms) {
if (mReceiver.getAudioTo(output, options).mStatus != AudioReceiver::DecodeResult::Status::Ok)
break;
} else
break;
}
// if (output.filled() < needed)
// ICELogError(<< "Not enough data for speaker's mixer");
}