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mythraopconnection.cpp
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mythraopconnection.cpp
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#include <QTimer>
#include <QTcpSocket>
#include <QtEndian>
#include <QTextStream>
#include "mythlogging.h"
#include "mythcorecontext.h"
#include "mythdirs.h"
#include "serverpool.h"
#include "audiooutput.h"
#include "audiooutpututil.h"
#include "mythraopdevice.h"
#include "mythraopconnection.h"
#include "mythairplayserver.h"
#include "mythuinotificationcenter.h"
#define LOC QString("RAOP Conn: ")
#define MAX_PACKET_SIZE 2048
RSA *MythRAOPConnection::g_rsa = NULL;
QString MythRAOPConnection::g_rsaLastError;
// RAOP RTP packet type
#define TIMING_REQUEST 0x52
#define TIMING_RESPONSE 0x53
#define SYNC 0x54
#define FIRSTSYNC (0x54 | 0x80)
#define RANGE_RESEND 0x55
#define AUDIO_RESEND 0x56
#define AUDIO_DATA 0x60
#define FIRSTAUDIO_DATA (0x60 | 0x80)
// Size (in ms) of audio buffered in audio card
#define AUDIOCARD_BUFFER 500
// How frequently we may call ProcessAudio (via QTimer)
// ideally 20ms, but according to documentation
// anything lower than 50ms on windows, isn't reliable
#define AUDIO_BUFFER 100
class _NetStream : public QTextStream
{
public:
_NetStream(QIODevice *device) : QTextStream(device)
{
};
_NetStream &operator<<(const QString &str)
{
LOG(VB_PLAYBACK, LOG_DEBUG,
LOC + QString("Sending(%1): ").arg(str.length()) + str);
QTextStream *q = this;
*q << str;
return *this;
};
};
MythRAOPConnection::MythRAOPConnection(QObject *parent, QTcpSocket *socket,
QByteArray id, int port)
: QObject(parent), m_watchdogTimer(NULL), m_socket(socket),
m_textStream(NULL), m_hardwareId(id),
m_incomingHeaders(), m_incomingContent(), m_incomingPartial(false),
m_incomingSize(0),
m_dataSocket(NULL), m_dataPort(port),
m_clientControlSocket(NULL), m_clientControlPort(0),
m_clientTimingSocket(NULL), m_clientTimingPort(0),
m_eventServer(NULL),
m_audio(NULL), m_codec(NULL), m_codeccontext(NULL),
m_channels(2), m_sampleSize(16), m_frameRate(44100),
m_framesPerPacket(352),m_dequeueAudioTimer(NULL),
m_queueLength(0), m_streamingStarted(false),
m_allowVolumeControl(true),
//audio sync
m_seqNum(0),
m_lastSequence(0), m_lastTimestamp(0),
m_currentTimestamp(0), m_nextSequence(0), m_nextTimestamp(0),
m_bufferLength(0), m_timeLastSync(0),
m_cardLatency(-1), m_adjustedLatency(-1), m_audioStarted(false),
// clock sync
m_masterTimeStamp(0), m_deviceTimeStamp(0), m_networkLatency(0),
m_clockSkew(0),
m_audioTimer(NULL),
m_progressStart(0), m_progressCurrent(0), m_progressEnd(0),
m_firstsend(false), m_playbackStarted(false)
{
m_id = MythUINotificationCenter::GetInstance()->Register(this);
}
MythRAOPConnection::~MythRAOPConnection()
{
CleanUp();
foreach (QTcpSocket *client, m_eventClients)
{
client->close();
client->deleteLater();
}
m_eventClients.clear();
if (m_eventServer)
{
m_eventServer->disconnect();
m_eventServer->close();
m_eventServer->deleteLater();
m_eventServer = NULL;
}
// delete main socket
if (m_socket)
{
m_socket->disconnect();
m_socket->close();
m_socket->deleteLater();
m_socket = NULL;
}
if (m_textStream)
{
delete m_textStream;
m_textStream = NULL;
}
if (m_id > 0)
{
MythUINotificationCenter::GetInstance()->UnRegister(this, m_id);
}
}
void MythRAOPConnection::CleanUp(void)
{
// delete audio timer
StopAudioTimer();
// stop and delete watchdog timer
if (m_watchdogTimer)
{
m_watchdogTimer->stop();
delete m_watchdogTimer;
m_watchdogTimer = NULL;
}
if (m_dequeueAudioTimer)
{
m_dequeueAudioTimer->stop();
delete m_dequeueAudioTimer;
m_dequeueAudioTimer = NULL;
}
if (m_clientTimingSocket)
{
m_clientTimingSocket->disconnect();
m_clientTimingSocket->close();
delete m_clientTimingSocket;
m_clientTimingSocket = NULL;
}
// delete data socket
if (m_dataSocket)
{
m_dataSocket->disconnect();
m_dataSocket->close();
m_dataSocket->deleteLater();
m_dataSocket = NULL;
}
// client control socket
if (m_clientControlSocket)
{
m_clientControlSocket->disconnect();
m_clientControlSocket->close();
m_clientControlSocket->deleteLater();
m_clientControlSocket = NULL;
}
// close audio decoder
DestroyDecoder();
// free decoded audio buffer
ResetAudio();
// close audio device
CloseAudioDevice();
// Tell listeners we're done
if (m_playbackStarted)
{
gCoreContext->emitTVPlaybackStopped();
}
}
bool MythRAOPConnection::Init(void)
{
// connect up the request socket
m_textStream = new _NetStream(m_socket);
m_textStream->setCodec("UTF-8");
if (!connect(m_socket, SIGNAL(readyRead()), this, SLOT(readClient())))
{
LOG(VB_PLAYBACK, LOG_ERR, LOC + "Failed to connect client socket signal.");
return false;
}
// create the data socket
m_dataSocket = new ServerPool();
if (!connect(m_dataSocket, SIGNAL(newDatagram(QByteArray, QHostAddress, quint16)),
this, SLOT(udpDataReady(QByteArray, QHostAddress, quint16))))
{
LOG(VB_PLAYBACK, LOG_ERR, LOC + "Failed to connect data socket signal.");
return false;
}
// try a few ports in case the first is in use
m_dataPort = m_dataSocket->tryBindingPort(m_dataPort, RAOP_PORT_RANGE);
if (m_dataPort < 0)
{
LOG(VB_PLAYBACK, LOG_ERR, LOC + "Failed to bind to a port for data.");
return false;
}
LOG(VB_PLAYBACK, LOG_INFO, LOC +
QString("Bound to port %1 for incoming data").arg(m_dataPort));
// load the private key
if (!LoadKey())
return false;
// use internal volume control
m_allowVolumeControl = gCoreContext->GetNumSetting("MythControlsVolume", 1);
// start the watchdog timer to auto delete the client after a period of inactivity
m_watchdogTimer = new QTimer();
connect(m_watchdogTimer, SIGNAL(timeout()), this, SLOT(timeout()));
m_watchdogTimer->start(10000);
m_dequeueAudioTimer = new QTimer();
connect(m_dequeueAudioTimer, SIGNAL(timeout()), this, SLOT(ProcessAudio()));
return true;
}
/**
* Socket incoming data signal handler
* use for audio, control and timing socket
*/
void MythRAOPConnection::udpDataReady(QByteArray buf, QHostAddress peer,
quint16 port)
{
// restart the idle timer
if (m_watchdogTimer)
m_watchdogTimer->start(10000);
if (!m_audio || !m_codec || !m_codeccontext)
return;
uint8_t type;
uint16_t seq;
uint64_t timestamp;
if (!GetPacketType(buf, type, seq, timestamp))
{
LOG(VB_PLAYBACK, LOG_DEBUG, LOC +
QString("Packet doesn't start with valid Rtp Header (0x%1)")
.arg((uint8_t)buf[0], 0, 16));
return;
}
switch (type)
{
case SYNC:
case FIRSTSYNC:
ProcessSync(buf);
ProcessAudio();
return;
case FIRSTAUDIO_DATA:
m_nextSequence = seq;
m_nextTimestamp = timestamp;
// With iTunes we know what the first sequence is going to be.
// iOS device do not tell us before streaming start what the first
// packet is going to be.
m_streamingStarted = true;
break;
case AUDIO_DATA:
case AUDIO_RESEND:
break;
case TIMING_RESPONSE:
ProcessTimeResponse(buf);
return;
default:
LOG(VB_PLAYBACK, LOG_DEBUG, LOC +
QString("Packet type (0x%1) not handled")
.arg(type, 0, 16));
return;
}
timestamp = framesToMs(timestamp);
if (timestamp < m_currentTimestamp)
{
LOG(VB_PLAYBACK, LOG_DEBUG, LOC +
QString("Received packet %1 too late, ignoring")
.arg(seq));
return;
}
// regular data packet
if (type == AUDIO_DATA || type == FIRSTAUDIO_DATA)
{
if (m_streamingStarted && seq != m_nextSequence)
SendResendRequest(timestamp, m_nextSequence, seq);
m_nextSequence = seq + 1;
m_nextTimestamp = timestamp;
m_streamingStarted = true;
}
if (!m_streamingStarted)
return;
// resent packet
if (type == AUDIO_RESEND)
{
if (m_resends.contains(seq))
{
LOG(VB_PLAYBACK, LOG_DEBUG, LOC +
QString("Received required resend %1 (with ts:%2 last:%3)")
.arg(seq).arg(timestamp).arg(m_nextSequence));
m_resends.remove(seq);
}
else
LOG(VB_PLAYBACK, LOG_WARNING, LOC +
QString("Received unexpected resent packet %1")
.arg(seq));
}
// Check that the audio packet is valid, do so by decoding it. If an error
// occurs, ask to resend it
QList<AudioData> *decoded = new QList<AudioData>();
int numframes = decodeAudioPacket(type, &buf, decoded);
if (numframes < 0)
{
// an error occurred, ask for the audio packet once again.
LOG(VB_PLAYBACK, LOG_ERR, LOC + QString("Error decoding audio"));
SendResendRequest(timestamp, seq, seq+1);
return;
}
AudioPacket frames;
frames.seq = seq;
frames.data = decoded;
m_audioQueue.insert(timestamp, frames);
ProcessAudio();
}
void MythRAOPConnection::ProcessSync(const QByteArray &buf)
{
bool first = (uint8_t)buf[0] == 0x90; // First sync is 0x90,0x55
const char *req = buf.constData();
uint64_t current_ts = qFromBigEndian(*(uint32_t *)(req + 4));
uint64_t next_ts = qFromBigEndian(*(uint32_t *)(req + 16));
uint64_t current = framesToMs(current_ts);
uint64_t next = framesToMs(next_ts);
m_currentTimestamp = current;
m_nextTimestamp = next;
m_bufferLength = m_nextTimestamp - m_currentTimestamp;
if (current_ts > m_progressStart)
{
m_progressCurrent = next_ts;
SendNotification(true);
}
if (first)
{
LOG(VB_PLAYBACK, LOG_DEBUG, LOC + QString("Receiving first SYNC packet"));
}
else
{
LOG(VB_PLAYBACK, LOG_DEBUG, LOC + QString("Receiving SYNC packet"));
}
timeval t; gettimeofday(&t, NULL);
m_timeLastSync = t.tv_sec * 1000 + t.tv_usec / 1000;
LOG(VB_PLAYBACK, LOG_DEBUG, LOC + QString("SYNC: cur:%1 next:%2 time:%3")
.arg(m_currentTimestamp).arg(m_nextTimestamp).arg(m_timeLastSync));
int64_t delay = framesToMs(m_audioQueue.size() * m_framesPerPacket);
int64_t audiots = m_audio->GetAudiotime();
int64_t currentLatency = 0LL;
if (m_audioStarted)
{
currentLatency = (int64_t)audiots - (int64_t)m_currentTimestamp;
}
LOG(VB_PLAYBACK, LOG_DEBUG, LOC +
QString("RAOP timestamps: about to play:%1 desired:%2 latency:%3")
.arg(audiots).arg(m_currentTimestamp)
.arg(currentLatency));
delay += m_audio->GetAudioBufferedTime();
delay += currentLatency;
LOG(VB_PLAYBACK, LOG_DEBUG, LOC +
QString("Queue=%1 buffer=%2ms ideal=%3ms diffts:%4ms")
.arg(m_audioQueue.size())
.arg(delay)
.arg(m_bufferLength)
.arg(m_bufferLength-delay));
if (m_adjustedLatency <= 0 && m_audioStarted &&
(-currentLatency > AUDIOCARD_BUFFER))
{
// Too much delay in playback.
// The threshold is a value chosen to be loose enough so it doesn't
// trigger too often, but should be low enough to detect any accidental
// interruptions.
// Will drop some frames in next ProcessAudio
LOG(VB_PLAYBACK, LOG_DEBUG, LOC +
QString("Too much delay (%1ms), adjusting")
.arg(m_bufferLength - delay));
m_adjustedLatency = m_cardLatency + m_networkLatency;
// Expire old audio
ExpireResendRequests(m_currentTimestamp - m_adjustedLatency);
int res = ExpireAudio(m_currentTimestamp - m_adjustedLatency);
if (res > 0)
{
LOG(VB_PLAYBACK, LOG_DEBUG, LOC + QString("Drop %1 packets").arg(res));
}
m_audioStarted = false;
}
}
/**
* SendResendRequest:
* Request RAOP client to resend missed RTP packets
*/
void MythRAOPConnection::SendResendRequest(uint64_t timestamp,
uint16_t expected, uint16_t got)
{
if (!m_clientControlSocket)
return;
int16_t missed = (got < expected) ?
(int16_t)(((int32_t)got + UINT16_MAX + 1) - expected) :
got - expected;
LOG(VB_PLAYBACK, LOG_INFO, LOC +
QString("Missed %1 packet(s): expected %2 got %3 ts:%4")
.arg(missed).arg(expected).arg(got).arg(timestamp));
char req[8];
req[0] = 0x80;
req[1] = RANGE_RESEND | 0x80;
*(uint16_t *)(req + 2) = qToBigEndian(m_seqNum++);
*(uint16_t *)(req + 4) = qToBigEndian(expected); // missed seqnum
*(uint16_t *)(req + 6) = qToBigEndian(missed); // count
if (m_clientControlSocket->writeDatagram(req, sizeof(req),
m_peerAddress, m_clientControlPort)
== sizeof(req))
{
for (uint16_t count = 0; count < missed; count++)
{
LOG(VB_PLAYBACK, LOG_INFO, LOC + QString("Sent resend for %1")
.arg(expected + count));
m_resends.insert(expected + count, timestamp);
}
}
else
LOG(VB_PLAYBACK, LOG_ERR, LOC + "Failed to send resend request.");
}
/**
* ExpireResendRequests:
* Expire resend requests that are older than timestamp. Those requests are
* expired when audio with older timestamp has already been played
*/
void MythRAOPConnection::ExpireResendRequests(uint64_t timestamp)
{
if (m_resends.isEmpty())
return;
QMutableMapIterator<uint16_t,uint64_t> it(m_resends);
while (it.hasNext())
{
it.next();
if (it.value() < timestamp && m_streamingStarted)
{
LOG(VB_PLAYBACK, LOG_WARNING, LOC +
QString("Never received resend packet %1").arg(it.key()));
m_resends.remove(it.key());
}
}
}
/**
* SendTimeRequest:
* Send a time request to the RAOP client.
*/
void MythRAOPConnection::SendTimeRequest(void)
{
if (!m_clientControlSocket) // should never happen
return;
timeval t;
gettimeofday(&t, NULL);
char req[32];
req[0] = 0x80;
req[1] = TIMING_REQUEST | 0x80;
// this is always 0x00 0x07 according to http://blog.technologeek.org/airtunes-v2
// no other value works
req[2] = 0x00;
req[3] = 0x07;
*(uint32_t *)(req + 4) = (uint32_t)0;
*(uint64_t *)(req + 8) = (uint64_t)0;
*(uint64_t *)(req + 16) = (uint64_t)0;
*(uint32_t *)(req + 24) = qToBigEndian((uint32_t)t.tv_sec);
*(uint32_t *)(req + 28) = qToBigEndian((uint32_t)t.tv_usec);
if (m_clientTimingSocket->writeDatagram(req, sizeof(req), m_peerAddress, m_clientTimingPort) != sizeof(req))
{
LOG(VB_PLAYBACK, LOG_ERR, LOC + "Failed to send resend time request.");
return;
}
LOG(VB_PLAYBACK, LOG_DEBUG, LOC +
QString("Requesting master time (Local %1.%2)")
.arg(t.tv_sec).arg(t.tv_usec));
}
/**
* ProcessTimeResponse:
* Calculate the network latency, we do not use the reference time send by itunes
* instead we measure the time lapsed between the request and the response
* the latency is calculated in ms
*/
void MythRAOPConnection::ProcessTimeResponse(const QByteArray &buf)
{
timeval t1, t2;
const char *req = buf.constData();
t1.tv_sec = qFromBigEndian(*(uint32_t *)(req + 8));
t1.tv_usec = qFromBigEndian(*(uint32_t *)(req + 12));
gettimeofday(&t2, NULL);
uint64_t time1, time2;
time1 = t1.tv_sec * 1000 + t1.tv_usec / 1000;
time2 = t2.tv_sec * 1000 + t2.tv_usec / 1000;
LOG(VB_PLAYBACK, LOG_DEBUG, LOC + QString("Read back time (Local %1.%2)")
.arg(t1.tv_sec).arg(t1.tv_usec));
// network latency equal time difference in ms between request and response
// divide by two for approximate time of one way trip
m_networkLatency = (time2 - time1) / 2;
LOG(VB_AUDIO, LOG_DEBUG, LOC + QString("Network Latency: %1ms")
.arg(m_networkLatency));
// now calculate the time difference between the client and us.
// this is NTP time, where sec is in seconds, and ticks is in 1/2^32s
uint32_t sec = qFromBigEndian(*(uint32_t *)(req + 24));
uint32_t ticks = qFromBigEndian(*(uint32_t *)(req + 28));
// convert ticks into ms
int64_t master = NTPToLocal(sec, ticks);
m_clockSkew = master - time2;
}
uint64_t MythRAOPConnection::NTPToLocal(uint32_t sec, uint32_t ticks)
{
return (int64_t)sec * 1000LL + (((int64_t)ticks * 1000LL) >> 32);
}
bool MythRAOPConnection::GetPacketType(const QByteArray &buf, uint8_t &type,
uint16_t &seq, uint64_t ×tamp)
{
// All RAOP packets start with | 0x80/0x90 (first sync) | PACKET_TYPE |
if ((uint8_t)buf[0] != 0x80 && (uint8_t)buf[0] != 0x90)
{
return false;
}
type = (char)buf[1];
// Is it first sync packet?
if ((uint8_t)buf[0] == 0x90 && type == FIRSTSYNC)
{
return true;
}
if (type != FIRSTAUDIO_DATA)
{
type &= ~0x80;
}
if (type != AUDIO_DATA && type != FIRSTAUDIO_DATA && type != AUDIO_RESEND)
return true;
const char *ptr = buf.constData();
if (type == AUDIO_RESEND)
{
ptr += 4;
}
seq = qFromBigEndian(*(uint16_t *)(ptr + 2));
timestamp = qFromBigEndian(*(uint32_t *)(ptr + 4));
return true;
}
// Audio decode / playback related routines
uint32_t MythRAOPConnection::decodeAudioPacket(uint8_t type,
const QByteArray *buf,
QList<AudioData> *dest)
{
const char *data_in = buf->constData();
int len = buf->size();
if (type == AUDIO_RESEND)
{
data_in += 4;
len -= 4;
}
data_in += 12;
len -= 12;
if (len < 16)
return -1;
int aeslen = len & ~0xf;
unsigned char iv[16];
unsigned char decrypted_data[MAX_PACKET_SIZE];
memcpy(iv, m_AESIV.constData(), sizeof(iv));
AES_cbc_encrypt((const unsigned char *)data_in,
decrypted_data, aeslen,
&m_aesKey, iv, AES_DECRYPT);
memcpy(decrypted_data + aeslen, data_in + aeslen, len - aeslen);
AVPacket tmp_pkt;
AVCodecContext *ctx = m_codeccontext;
av_init_packet(&tmp_pkt);
tmp_pkt.data = decrypted_data;
tmp_pkt.size = len;
uint32_t frames_added = 0;
uint8_t *samples = (uint8_t *)av_mallocz(AVCODEC_MAX_AUDIO_FRAME_SIZE);
while (tmp_pkt.size > 0)
{
int data_size;
int ret = AudioOutputUtil::DecodeAudio(ctx, samples,
data_size, &tmp_pkt);
if (ret < 0)
{
av_free(samples);
return -1;
}
if (data_size)
{
int num_samples = data_size /
(ctx->channels * av_get_bytes_per_sample(ctx->sample_fmt));
frames_added += num_samples;
AudioData block;
block.data = samples;
block.length = data_size;
block.frames = num_samples;
dest->append(block);
}
tmp_pkt.data += ret;
tmp_pkt.size -= ret;
}
return frames_added;
}
void MythRAOPConnection::ProcessAudio()
{
if (!m_streamingStarted || !m_audio)
return;
if (m_audio->IsPaused())
{
// ALSA takes a while to unpause, enough to have SYNC starting to drop
// packets, so unpause as early as possible
m_audio->Pause(false);
}
timeval t; gettimeofday(&t, NULL);
uint64_t dtime = (t.tv_sec * 1000 + t.tv_usec / 1000) - m_timeLastSync;
uint64_t rtp = dtime + m_currentTimestamp;
uint64_t buffered = m_audioStarted ? m_audio->GetAudioBufferedTime() : 0;
// Keep audio framework buffer as short as possible, keeping everything in
// m_audioQueue, so we can easily reset the least amount possible
if (buffered > AUDIOCARD_BUFFER)
return;
// Also make sure m_audioQueue never goes to less than 1/3 of the RDP stream
// total latency, this should gives us enough time to receive missed packets
int64_t queue = framesToMs(m_audioQueue.size() * m_framesPerPacket);
if (queue < m_bufferLength / 3)
return;
rtp += buffered;
// How many packets to add to the audio card, to fill AUDIOCARD_BUFFER
int max_packets = ((AUDIOCARD_BUFFER - buffered)
* m_frameRate / 1000) / m_framesPerPacket;
int i = 0;
uint64_t timestamp = 0;
QMapIterator<uint64_t,AudioPacket> packet_it(m_audioQueue);
while (packet_it.hasNext() && i <= max_packets)
{
packet_it.next();
timestamp = packet_it.key();
if (timestamp < rtp)
{
if (!m_audioStarted)
{
m_audio->Reset(); // clear audio card
}
AudioPacket frames = packet_it.value();
if (m_lastSequence != frames.seq)
{
LOG(VB_PLAYBACK, LOG_ERR, LOC +
QString("Audio discontinuity seen. Played %1 (%3) expected %2")
.arg(frames.seq).arg(m_lastSequence).arg(timestamp));
m_lastSequence = frames.seq;
}
m_lastSequence++;
QList<AudioData>::iterator it = frames.data->begin();
for (; it != frames.data->end(); ++it)
{
AudioData *data = &(*it);
int offset = 0;
int frames = 0;
if (m_adjustedLatency > 0)
{
// calculate how many frames we have to drop to catch up
offset = (m_adjustedLatency * m_frameRate / 1000) *
m_audio->GetBytesPerFrame();
if (offset > data->length)
offset = data->length;
frames = offset / m_audio->GetBytesPerFrame();
m_adjustedLatency -= framesToMs(frames+1);
LOG(VB_PLAYBACK, LOG_DEBUG, LOC +
QString("ProcessAudio: Dropping %1 frames to catch up "
"(%2ms to go)")
.arg(frames).arg(m_adjustedLatency));
timestamp += framesToMs(frames);
}
m_audio->AddData((char *)data->data + offset,
data->length - offset,
timestamp, frames);
timestamp += m_audio->LengthLastData();
}
i++;
m_audioStarted = true;
}
else // QMap is sorted, so no need to continue if not found
break;
}
ExpireAudio(timestamp);
m_lastTimestamp = timestamp;
// restart audio timer should we stop receiving data on regular interval,
// we need to continue processing the audio queue
m_dequeueAudioTimer->start(AUDIO_BUFFER);
}
int MythRAOPConnection::ExpireAudio(uint64_t timestamp)
{
int res = 0;
QMutableMapIterator<uint64_t,AudioPacket> packet_it(m_audioQueue);
while (packet_it.hasNext())
{
packet_it.next();
if (packet_it.key() < timestamp)
{
AudioPacket frames = packet_it.value();
if (frames.data)
{
QList<AudioData>::iterator it = frames.data->begin();
for (; it != frames.data->end(); ++it)
{
av_free(it->data);
}
delete frames.data;
}
m_audioQueue.remove(packet_it.key());
res++;
}
}
return res;
}
void MythRAOPConnection::ResetAudio(void)
{
if (m_audio)
{
m_audio->Reset();
}
ExpireAudio(UINT64_MAX);
ExpireResendRequests(UINT64_MAX);
m_audioStarted = false;
}
void MythRAOPConnection::timeout(void)
{
LOG(VB_PLAYBACK, LOG_INFO, LOC + "Closing connection after inactivity.");
m_socket->disconnectFromHost();
}
void MythRAOPConnection::audioRetry(void)
{
if (!m_audio && OpenAudioDevice())
{
CreateDecoder();
}
if (m_audio && m_codec && m_codeccontext)
{
StopAudioTimer();
}
}
/**
* readClient: signal handler for RAOP client connection
* Handle initialisation of session
*/
void MythRAOPConnection::readClient(void)
{
QTcpSocket *socket = (QTcpSocket *)sender();
if (!socket)
return;
QByteArray data = socket->readAll();
LOG(VB_PLAYBACK, LOG_DEBUG, LOC + QString("readClient(%1): ")
.arg(data.size()) + data.constData());
// For big content, we may be called several times for a single packet
if (!m_incomingPartial)
{
m_incomingHeaders.clear();
m_incomingContent.clear();
m_incomingSize = 0;
QTextStream stream(data);
QString line;
do
{
line = stream.readLine();
if (line.size() == 0)
break;
LOG(VB_PLAYBACK, LOG_DEBUG, LOC + QString("Header(%1) = %2")
.arg(m_socket->peerAddress().toString())
.arg(line));
m_incomingHeaders.append(line);
if (line.contains("Content-Length:"))
{
m_incomingSize = line.mid(line.indexOf(" ") + 1).toInt();
}
}
while (!line.isNull());
if (m_incomingHeaders.size() == 0)
return;
if (!stream.atEnd())
{
int pos = stream.pos();
if (pos > 0)
{
m_incomingContent.append(data.mid(pos));
}
}
}
else
{
m_incomingContent.append(data);
}
// If we haven't received all the content yet, wait (see when receiving
// coverart
if (m_incomingContent.size() < m_incomingSize)
{
m_incomingPartial = true;
return;
}
else
{
m_incomingPartial = false;
}
LOG(VB_PLAYBACK, LOG_DEBUG, LOC + QString("Content(%1) = %2")
.arg(m_incomingContent.size()).arg(m_incomingContent.constData()));
ProcessRequest(m_incomingHeaders, m_incomingContent);
}
void MythRAOPConnection::ProcessRequest(const QStringList &header,
const QByteArray &content)
{
if (header.isEmpty())
return;
RawHash tags = FindTags(header);
if (!tags.contains("CSeq"))
{
LOG(VB_PLAYBACK, LOG_ERR, LOC + "ProcessRequest: Didn't find CSeq");
return;
}
QString option = header[0].left(header[0].indexOf(" "));
// process RTP-info field
bool gotRTP = false;
uint16_t RTPseq;
uint64_t RTPtimestamp;
if (tags.contains("RTP-Info"))
{
gotRTP = true;
QString data = tags["RTP-Info"];
QStringList items = data.split(";");
foreach (QString item, items)
{
if (item.startsWith("seq"))
{
RTPseq = item.mid(item.indexOf("=") + 1).trimmed().toUShort();
}
else if (item.startsWith("rtptime"))
{
RTPtimestamp = item.mid(item.indexOf("=") + 1).trimmed().toUInt();
}
}
LOG(VB_PLAYBACK, LOG_INFO, LOC + QString("RTP-Info: seq=%1 rtptime=%2")
.arg(RTPseq).arg(RTPtimestamp));
}
if (gCoreContext->GetNumSetting("AirPlayPasswordEnabled", false))
{
if (m_nonce.isEmpty())
{
m_nonce = GenerateNonce();
}
if (!tags.contains("Authorization"))
{
// 60 seconds to enter password.
m_watchdogTimer->start(60000);
FinishAuthenticationResponse(m_textStream, m_socket, tags["CSeq"]);
return;
}
QByteArray auth;
if (DigestMd5Response(tags["Authorization"], option, m_nonce,
gCoreContext->GetSetting("AirPlayPassword"),
auth) == auth)
{
LOG(VB_PLAYBACK, LOG_INFO, LOC + "RAOP client authenticated");
}
else
{
LOG(VB_PLAYBACK, LOG_INFO, LOC + "RAOP authentication failed");
FinishAuthenticationResponse(m_textStream, m_socket, tags["CSeq"]);
return;
}
}
*m_textStream << "RTSP/1.0 200 OK\r\n";
if (tags.contains("Apple-Challenge"))
{
LOG(VB_PLAYBACK, LOG_DEBUG, LOC + QString("Received Apple-Challenge"));
*m_textStream << "Apple-Response: ";
if (!LoadKey())
return;
int tosize = RSA_size(LoadKey());
uint8_t *to = new uint8_t[tosize];
QByteArray challenge =
QByteArray::fromBase64(tags["Apple-Challenge"].toLatin1());
int challenge_size = challenge.size();
if (challenge_size != 16)
{
LOG(VB_PLAYBACK, LOG_ERR, LOC +
QString("Decoded challenge size %1, expected 16")
.arg(challenge_size));
if (challenge_size > 16)
challenge_size = 16;
}
int i = 0;
unsigned char from[38];
memcpy(from, challenge.constData(), challenge_size);
i += challenge_size;