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Copy audio to UDP/RTP: Opus implementation (4)

pull/295/head
f4exb 2019-02-19 00:36:32 +01:00
rodzic 297dcce2d3
commit 0aaab42f95
2 zmienionych plików z 74 dodań i 3 usunięć
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68
sdrbase/audio/audionetsink.cpp
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@ -269,6 +269,19 @@ void AudioNetSink::write(qint16 isample)
}
}
break;
case CodecOpus:
{
if (m_codecInputIndex == m_codecInputSize)
{
int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
m_codecInputIndex = 0;
}
m_opusIn[m_codecInputIndex++] = sample;
}
break;
case CodecL16:
default:
{
@ -314,6 +327,25 @@ void AudioNetSink::write(qint16 isample)
m_bufferIndex += 1;
}
break;
case CodecOpus:
{
if (m_codecInputIndex == m_codecInputSize)
{
int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
qWarning("AudioNetSink::write: CodecOpus mono: unexpected output frame size: %d bytes", nbBytes);
}
m_bufferIndex = 0;
m_codecInputIndex = 0;
}
if (m_codecInputIndex % m_codecRatio == 0) {
m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
}
m_opusIn[m_codecInputIndex++] = sample;
}
break;
case CodecL16:
default:
m_rtpBufferAudio->write((uint8_t *) &sample);
@ -359,6 +391,21 @@ void AudioNetSink::write(qint16 ilSample, qint16 irSample)
case CodecPCMU:
case CodecG722:
break; // mono modes - do nothing
case CodecOpus:
{
if (m_codecInputIndex == m_codecInputSize)
{
int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
m_codecInputIndex = 0;
}
m_opusIn[2*m_codecInputIndex] = lSample;
m_opusIn[2*m_codecInputIndex+1] = rSample;
m_codecInputIndex++;
}
break;
case CodecL8:
{
qint8 *p = (qint8*) &m_data[m_bufferIndex];
@ -390,6 +437,27 @@ void AudioNetSink::write(qint16 ilSample, qint16 irSample)
case CodecPCMU:
case CodecG722:
break; // mono modes - do nothing
case CodecOpus:
{
if (m_codecInputIndex == m_codecInputSize)
{
int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
qWarning("AudioNetSink::write: CodecOpus stereo: unexpected output frame size: %d bytes", nbBytes);
}
m_bufferIndex = 0;
m_codecInputIndex = 0;
}
if (m_codecInputIndex % m_codecRatio == 0) {
m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
}
m_opusIn[2*m_codecInputIndex] = lSample;
m_opusIn[2*m_codecInputIndex+1] = rSample;
m_codecInputIndex++;
}
break;
case CodecL8:
{
qint8 pl = lSample / 256;

9
sdrbase/util/rtpsink.cpp
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@ -112,9 +112,9 @@ void RTPSink::setPayloadInformation(PayloadType payloadType, int sampleRate)
break;
case PayloadOpus:
m_sampleBytes = 1;
m_rtpSession.SetDefaultPayloadType(101);
m_packetSamples = 960; // Fixed 20ms @ 48 kHz as per https://tools.ietf.org/html/rfc7587
timestampinc = 960; // and per single channel
m_rtpSession.SetDefaultPayloadType(96);
m_packetSamples = 160; // Payload size is 160 bytes
timestampinc = 960; // But increment is 960
break;
case PayloadL16Mono:
default:
@ -336,6 +336,9 @@ unsigned int RTPSink::elemLength(PayloadType payloadType)
{
case PayloadPCMA8:
case PayloadPCMU8:
case PayloadG722:
case PayloadOpus:
case PayloadL8:
return sizeof(int8_t);
break;
case PayloadL16Stereo: