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wireshark/ui/qt/rtp_audio_stream.cpp

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/* rtp_audio_frame.cpp
*
* Wireshark - Network traffic analyzer
* By Gerald Combs <gerald@wireshark.org>
* Copyright 1998 Gerald Combs
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "rtp_audio_stream.h"
#ifdef QT_MULTIMEDIA_LIB
#ifdef HAVE_SPEEXDSP
#include <speex/speex_resampler.h>
#else
#include <codecs/speex/speex_resampler.h>
#endif /* HAVE_SPEEXDSP */
#include <epan/rtp_pt.h>
#include <epan/dissectors/packet-rtp.h>
#include <ui/rtp_media.h>
#include <ui/rtp_stream.h>
#include <wsutil/nstime.h>
#include <QAudioFormat>
#include <QAudioOutput>
#include <QDir>
#include <QTemporaryFile>
#include <QVariant>
// To do:
// - Only allow one rtp_stream_info_t per RtpAudioStream?
static spx_int16_t default_audio_sample_rate_ = 8000;
static const spx_int16_t visual_sample_rate_ = 1000;
RtpAudioStream::RtpAudioStream(QObject *parent, _rtp_stream_info *rtp_stream) :
QObject(parent),
decoders_hash_(rtp_decoder_hash_table_new()),
global_start_rel_time_(0.0),
start_abs_offset_(0.0),
start_rel_time_(0.0),
stop_rel_time_(0.0),
audio_out_rate_(0),
audio_resampler_(0),
audio_output_(0),
max_sample_val_(1),
color_(0),
jitter_buffer_size_(50),
timing_mode_(RtpAudioStream::JitterBuffer)
{
copy_address(&src_addr_, &rtp_stream->src_addr);
src_port_ = rtp_stream->src_port;
copy_address(&dst_addr_, &rtp_stream->dest_addr);
dst_port_ = rtp_stream->dest_port;
ssrc_ = rtp_stream->ssrc;
// We keep visual samples in memory. Make fewer of them.
visual_resampler_ = speex_resampler_init(1, default_audio_sample_rate_,
visual_sample_rate_, SPEEX_RESAMPLER_QUALITY_MIN, NULL);
speex_resampler_skip_zeros(visual_resampler_);
QString tempname = QString("%1/wireshark_rtp_stream").arg(QDir::tempPath());
tempfile_ = new QTemporaryFile(tempname, this);
tempfile_->open();
// RTP_STREAM_DEBUG("Writing to %s", tempname.toUtf8().constData());
}
RtpAudioStream::~RtpAudioStream()
{
for (int i = 0; i < rtp_packets_.size(); i++) {
rtp_packet_t *rtp_packet = rtp_packets_[i];
g_free(rtp_packet->info);
g_free(rtp_packet->payload_data);
g_free(rtp_packet);
}
g_hash_table_destroy(decoders_hash_);
if (audio_resampler_) speex_resampler_destroy (audio_resampler_);
speex_resampler_destroy (visual_resampler_);
}
bool RtpAudioStream::isMatch(const _rtp_stream_info *rtp_stream) const
{
if (rtp_stream
&& addresses_equal(&rtp_stream->src_addr, &src_addr_)
&& rtp_stream->src_port == src_port_
&& addresses_equal(&rtp_stream->dest_addr, &dst_addr_)
&& rtp_stream->dest_port == dst_port_
&& rtp_stream->ssrc == ssrc_)
return true;
return false;
}
bool RtpAudioStream::isMatch(const _packet_info *pinfo, const _rtp_info *rtp_info) const
{
if (pinfo && rtp_info
&& addresses_equal(&pinfo->src, &src_addr_)
&& pinfo->srcport == src_port_
&& addresses_equal(&pinfo->dst, &dst_addr_)
&& pinfo->destport == dst_port_
&& rtp_info->info_sync_src == ssrc_)
return true;
return false;
}
// XXX We add multiple RTP streams here because that's what the GTK+ UI does.
// Should we make these distinct, with their own waveforms? It seems like
// that would simplify a lot of things.
void RtpAudioStream::addRtpStream(const _rtp_stream_info *rtp_stream)
{
if (!rtp_stream) return;
// RTP_STREAM_DEBUG("added %d:%u packets", g_list_length(rtp_stream->rtp_packet_list), rtp_stream->packet_count);
rtp_streams_ << rtp_stream;
}
void RtpAudioStream::addRtpPacket(const struct _packet_info *pinfo, const struct _rtp_info *rtp_info)
{
// gtk/rtp_player.c:decode_rtp_packet
if (!rtp_info) return;
rtp_packet_t *rtp_packet = g_new0(rtp_packet_t, 1);
rtp_packet->info = (struct _rtp_info *) g_memdup(rtp_info, sizeof(struct _rtp_info));
if (rtp_info->info_all_data_present && (rtp_info->info_payload_len != 0)) {
rtp_packet->payload_data = (guint8 *) g_memdup(&(rtp_info->info_data[rtp_info->info_payload_offset]), rtp_info->info_payload_len);
}
if (rtp_packets_.size() < 1) { // First packet
start_abs_offset_ = nstime_to_sec(&pinfo->abs_ts) - start_rel_time_;
start_rel_time_ = stop_rel_time_ = nstime_to_sec(&pinfo->rel_ts);
}
rtp_packet->frame_num = pinfo->num;
rtp_packet->arrive_offset = nstime_to_sec(&pinfo->rel_ts) - start_rel_time_;
rtp_packets_ << rtp_packet;
}
void RtpAudioStream::reset(double start_rel_time)
{
global_start_rel_time_ = start_rel_time;
stop_rel_time_ = start_rel_time_;
audio_out_rate_ = 0;
max_sample_val_ = 1;
packet_timestamps_.clear();
visual_samples_.clear();
out_of_seq_timestamps_.clear();
jitter_drop_timestamps_.clear();
if (audio_resampler_) {
speex_resampler_reset_mem(audio_resampler_);
}
if (visual_resampler_) {
speex_resampler_reset_mem(visual_resampler_);
}
tempfile_->seek(0);
}
static const int sample_bytes_ = sizeof(SAMPLE) / sizeof(char);
/* Fix for bug 4119/5902: don't insert too many silence frames.
* XXX - is there a better thing to do here?
*/
static const int max_silence_samples_ = MAX_SILENCE_FRAMES;
void RtpAudioStream::decode()
{
if (rtp_packets_.size() < 1) return;
// gtk/rtp_player.c:decode_rtp_stream
// XXX This is more messy than it should be.
gsize resample_buff_len = 0x1000;
SAMPLE *resample_buff = (SAMPLE *) g_malloc(resample_buff_len);
spx_uint32_t cur_in_rate = 0, visual_out_rate = 0;
char *write_buff = NULL;
qint64 write_bytes = 0;
unsigned channels = 0;
unsigned sample_rate = 0;
int last_sequence = 0;
double rtp_time_prev = 0.0;
double arrive_time_prev = 0.0;
double pack_period = 0.0;
double start_time = 0.0;
double start_rtp_time = 0.0;
guint32 start_timestamp = 0;
size_t decoded_bytes_prev = 0;
for (int cur_packet = 0; cur_packet < rtp_packets_.size(); cur_packet++) {
SAMPLE *decode_buff = NULL;
// XXX The GTK+ UI updates a progress bar here.
rtp_packet_t *rtp_packet = rtp_packets_[cur_packet];
stop_rel_time_ = start_rel_time_ + rtp_packet->arrive_offset;
speex_resampler_get_rate(visual_resampler_, &cur_in_rate, &visual_out_rate);
QString payload_name;
if (rtp_packet->info->info_payload_type_str) {
payload_name = rtp_packet->info->info_payload_type_str;
} else {
payload_name = try_val_to_str_ext(rtp_packet->info->info_payload_type, &rtp_payload_type_short_vals_ext);
}
if (!payload_name.isEmpty()) {
payload_names_ << payload_name;
}
if (cur_packet < 1) { // First packet
start_timestamp = rtp_packet->info->info_timestamp;
start_rtp_time = 0;
rtp_time_prev = 0;
last_sequence = rtp_packet->info->info_seq_num - 1;
}
size_t decoded_bytes = decode_rtp_packet(rtp_packet, &decode_buff, decoders_hash_, &channels, &sample_rate);
if (decoded_bytes == 0 || sample_rate == 0) {
// We didn't decode anything. Clean up and prep for the next packet.
last_sequence = rtp_packet->info->info_seq_num;
g_free(decode_buff);
continue;
}
if (audio_out_rate_ == 0) { // First non-zero wins
audio_out_rate_ = sample_rate;
RTP_STREAM_DEBUG("Audio sample rate is %u", audio_out_rate_);
// Prepend silence to match our sibling streams.
tempfile_->seek(0);
int prepend_samples = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_;
if (prepend_samples > 0) {
writeSilence(prepend_samples);
}
}
if (rtp_packet->info->info_seq_num != last_sequence+1) {
out_of_seq_timestamps_.append(stop_rel_time_);
}
last_sequence = rtp_packet->info->info_seq_num;
unsigned rtp_clock_rate = sample_rate;
if (rtp_packet->info->info_payload_type == PT_G722) {
// G.722 sample rate is 16kHz, but RTP clock rate is 8kHz for historic reasons.
rtp_clock_rate = 8000;
}
double rtp_time = (double)(rtp_packet->info->info_timestamp-start_timestamp)/rtp_clock_rate - start_rtp_time;
double arrive_time;
if (timing_mode_ == RtpTimestamp) {
arrive_time = rtp_time;
} else {
arrive_time = rtp_packet->arrive_offset - start_time;
}
double diff = qAbs(arrive_time - rtp_time);
if (diff*1000 > jitter_buffer_size_ && timing_mode_ != Uninterrupted) {
// rtp_player.c:628
jitter_drop_timestamps_.append(stop_rel_time_);
RTP_STREAM_DEBUG("Packet drop by jitter buffer exceeded %f > %d", diff*1000, jitter_buffer_size_);
/* if there was a silence period (more than two packetization period) resync the source */
if ((rtp_time - rtp_time_prev) > pack_period*2) {
int silence_samples;
RTP_STREAM_DEBUG("Resync...");
silence_samples = (int)((arrive_time - arrive_time_prev)*sample_rate - decoded_bytes_prev / sample_bytes_);
/* Fix for bug 4119/5902: don't insert too many silence frames.
* XXX - is there a better thing to do here?
*/
silence_samples = qMin(silence_samples, max_silence_samples_);
writeSilence(silence_samples);
silence_timestamps_.append(stop_rel_time_);
decoded_bytes_prev = 0;
/* defined start_timestmp to avoid overflow in timestamp. TODO: handle the timestamp correctly */
/* XXX: if timestamps (RTP) are missing/ignored try use packet arrive time only (see also "rtp_time") */
start_timestamp = rtp_packet->info->info_timestamp;
start_rtp_time = 0;
start_time = rtp_packet->arrive_offset;
rtp_time_prev = 0;
}
} else {
// rtp_player.c:664
/* Add silence if it is necessary */
int silence_samples;
if (timing_mode_ == Uninterrupted) {
silence_samples = 0;
} else {
silence_samples = (int)((rtp_time - rtp_time_prev)*sample_rate - decoded_bytes_prev / sample_bytes_);
}
if (silence_samples != 0) {
wrong_timestamp_timestamps_.append(stop_rel_time_);
}
if (silence_samples > 0) {
/* Fix for bug 4119/5902: don't insert too many silence frames.
* XXX - is there a better thing to do here?
*/
silence_samples = qMin(silence_samples, max_silence_samples_);
writeSilence(silence_samples);
silence_timestamps_.append(stop_rel_time_);
}
// XXX rtp_player.c:696 adds audio here.
rtp_time_prev = rtp_time;
pack_period = (double) decoded_bytes / sample_bytes_ / sample_rate;
decoded_bytes_prev = decoded_bytes;
arrive_time_prev = arrive_time;
}
// Write samples to our file.
write_buff = (char *) decode_buff;
write_bytes = decoded_bytes;
if (audio_out_rate_ != sample_rate) {
// Resample the audio to match our previous output rate.
if (!audio_resampler_) {
audio_resampler_ = speex_resampler_init(1, sample_rate, audio_out_rate_, 10, NULL);
speex_resampler_skip_zeros(audio_resampler_);
RTP_STREAM_DEBUG("Started resampling from %u to (out) %u Hz.", sample_rate, audio_out_rate_);
} else {
spx_uint32_t audio_out_rate;
speex_resampler_get_rate(audio_resampler_, &cur_in_rate, &audio_out_rate);
// Adjust rates if needed.
if (sample_rate != cur_in_rate) {
speex_resampler_set_rate(audio_resampler_, sample_rate, audio_out_rate);
speex_resampler_set_rate(visual_resampler_, sample_rate, visual_out_rate);
RTP_STREAM_DEBUG("Changed input rate from %u to %u Hz. Out is %u.", cur_in_rate, sample_rate, audio_out_rate_);
}
}
spx_uint32_t in_len = (spx_uint32_t)rtp_packet->info->info_payload_len;
spx_uint32_t out_len = (audio_out_rate_ * (spx_uint32_t)rtp_packet->info->info_payload_len / sample_rate) + (audio_out_rate_ % sample_rate != 0);
if (out_len * sample_bytes_ > resample_buff_len) {
while ((out_len * sample_bytes_ > resample_buff_len))
resample_buff_len *= 2;
resample_buff = (SAMPLE *) g_realloc(resample_buff, resample_buff_len);
}
speex_resampler_process_int(audio_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
write_buff = (char *) resample_buff;
write_bytes = out_len * sample_bytes_;
}
// Write the decoded, possibly-resampled audio to our temp file.
tempfile_->write(write_buff, write_bytes);
// Collect our visual samples.
spx_uint32_t in_len = (spx_uint32_t)rtp_packet->info->info_payload_len;
spx_uint32_t out_len = (visual_out_rate * in_len / sample_rate) + (visual_out_rate % sample_rate != 0);
if (out_len * sample_bytes_ > resample_buff_len) {
while ((out_len * sample_bytes_ > resample_buff_len))
resample_buff_len *= 2;
resample_buff = (SAMPLE *) g_realloc(resample_buff, resample_buff_len);
}
speex_resampler_process_int(visual_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
for (unsigned i = 0; i < out_len; i++) {
packet_timestamps_[stop_rel_time_ + (double) i / visual_out_rate] = rtp_packet->frame_num;
if (qAbs(resample_buff[i]) > max_sample_val_) max_sample_val_ = qAbs(resample_buff[i]);
visual_samples_.append(resample_buff[i]);
}
// Finally, write the resampled audio to our temp file and clean up.
g_free(decode_buff);
}
g_free(resample_buff);
}
const QStringList RtpAudioStream::payloadNames() const
{
QStringList payload_names = payload_names_.toList();
payload_names.sort();
return payload_names;
}
const QVector<double> RtpAudioStream::visualTimestamps(bool relative)
{
QVector<double> ts_keys = packet_timestamps_.keys().toVector();
if (relative) return ts_keys;
QVector<double> adj_timestamps;
for (int i = 0; i < ts_keys.size(); i++) {
adj_timestamps.append(ts_keys[i] + start_abs_offset_);
}
return adj_timestamps;
}
// Scale the height of the waveform (max_sample_val_) and adjust its Y
// offset so that they overlap slightly (stack_offset_).
// XXX This means that waveforms can be misleading with respect to relative
// amplitude. We might want to add a "global" max_sample_val_.
static const double stack_offset_ = G_MAXINT16 / 3;
const QVector<double> RtpAudioStream::visualSamples(int y_offset)
{
QVector<double> adj_samples;
double scaled_offset = y_offset * stack_offset_;
for (int i = 0; i < visual_samples_.size(); i++) {
adj_samples.append(((double)visual_samples_[i] * G_MAXINT16 / max_sample_val_) + scaled_offset);
}
return adj_samples;
}
const QVector<double> RtpAudioStream::outOfSequenceTimestamps(bool relative)
{
if (relative) return out_of_seq_timestamps_;
QVector<double> adj_timestamps;
for (int i = 0; i < out_of_seq_timestamps_.size(); i++) {
adj_timestamps.append(out_of_seq_timestamps_[i] + start_abs_offset_);
}
return adj_timestamps;
}
const QVector<double> RtpAudioStream::outOfSequenceSamples(int y_offset)
{
QVector<double> adj_samples;
double scaled_offset = y_offset * stack_offset_; // XXX Should be different for seq, jitter, wrong & silence
for (int i = 0; i < out_of_seq_timestamps_.size(); i++) {
adj_samples.append(scaled_offset);
}
return adj_samples;
}
const QVector<double> RtpAudioStream::jitterDroppedTimestamps(bool relative)
{
if (relative) return jitter_drop_timestamps_;
QVector<double> adj_timestamps;
for (int i = 0; i < jitter_drop_timestamps_.size(); i++) {
adj_timestamps.append(jitter_drop_timestamps_[i] + start_abs_offset_);
}
return adj_timestamps;
}
const QVector<double> RtpAudioStream::jitterDroppedSamples(int y_offset)
{
QVector<double> adj_samples;
double scaled_offset = y_offset * stack_offset_; // XXX Should be different for seq, jitter, wrong & silence
for (int i = 0; i < jitter_drop_timestamps_.size(); i++) {
adj_samples.append(scaled_offset);
}
return adj_samples;
}
const QVector<double> RtpAudioStream::wrongTimestampTimestamps(bool relative)
{
if (relative) return wrong_timestamp_timestamps_;
QVector<double> adj_timestamps;
for (int i = 0; i < wrong_timestamp_timestamps_.size(); i++) {
adj_timestamps.append(wrong_timestamp_timestamps_[i] + start_abs_offset_);
}
return adj_timestamps;
}
const QVector<double> RtpAudioStream::wrongTimestampSamples(int y_offset)
{
QVector<double> adj_samples;
double scaled_offset = y_offset * stack_offset_; // XXX Should be different for seq, jitter, wrong & silence
for (int i = 0; i < wrong_timestamp_timestamps_.size(); i++) {
adj_samples.append(scaled_offset);
}
return adj_samples;
}
const QVector<double> RtpAudioStream::insertedSilenceTimestamps(bool relative)
{
if (relative) return silence_timestamps_;
QVector<double> adj_timestamps;
for (int i = 0; i < silence_timestamps_.size(); i++) {
adj_timestamps.append(silence_timestamps_[i] + start_abs_offset_);
}
return adj_timestamps;
}
const QVector<double> RtpAudioStream::insertedSilenceSamples(int y_offset)
{
QVector<double> adj_samples;
double scaled_offset = y_offset * stack_offset_; // XXX Should be different for seq, jitter, wrong & silence
for (int i = 0; i < silence_timestamps_.size(); i++) {
adj_samples.append(scaled_offset);
}
return adj_samples;
}
quint32 RtpAudioStream::nearestPacket(double timestamp, bool is_relative)
{
if (packet_timestamps_.keys().count() < 1) return 0;
if (!is_relative) timestamp -= start_abs_offset_;
QMap<double, quint32>::const_iterator it = packet_timestamps_.lowerBound(timestamp);
if (it == packet_timestamps_.end()) return 0;
return it.value();
}
QAudio::State RtpAudioStream::outputState() const
{
if (!audio_output_) return QAudio::IdleState;
return audio_output_->state();
}
const QString RtpAudioStream::formatDescription(const QAudioFormat &format)
{
QString fmt_descr = QString("%1 Hz, ").arg(format.sampleRate());
switch (format.sampleType()) {
case QAudioFormat::SignedInt:
fmt_descr += "Int";
break;
case QAudioFormat::UnSignedInt:
fmt_descr += "UInt";
break;
case QAudioFormat::Float:
fmt_descr += "Float";
break;
default:
fmt_descr += "Unknown";
break;
}
fmt_descr += QString::number(format.sampleSize());
fmt_descr += format.byteOrder() == QAudioFormat::BigEndian ? "BE" : "LE";
return fmt_descr;
}
void RtpAudioStream::startPlaying()
{
if (audio_output_) return;
if (audio_out_rate_ == 0) {
emit playbackError(tr("RTP stream is empty or codec is unsupported."));
return;
}
QAudioDeviceInfo cur_out_device = QAudioDeviceInfo::defaultOutputDevice();
QString cur_out_name = parent()->property("currentOutputDeviceName").toString();
foreach (QAudioDeviceInfo out_device, QAudioDeviceInfo::availableDevices(QAudio::AudioOutput)) {
if (cur_out_name == out_device.deviceName()) {
cur_out_device = out_device;
}
}
QAudioFormat format;
format.setSampleRate(audio_out_rate_);
format.setSampleSize(sample_bytes_ * 8); // bits
format.setSampleType(QAudioFormat::SignedInt);
format.setChannelCount(1);
format.setCodec("audio/pcm");
// RTP_STREAM_DEBUG("playing %s %d samples @ %u Hz",
// tempfile_->fileName().toUtf8().constData(),
// (int) tempfile_->size(), audio_out_rate_);
if (!cur_out_device.isFormatSupported(format)) {
QString playback_error = tr("%1 does not support PCM at %2. Preferred format is %3")
.arg(cur_out_device.deviceName())
.arg(formatDescription(format))
.arg(formatDescription(cur_out_device.nearestFormat(format)));
emit playbackError(playback_error);
}
audio_output_ = new QAudioOutput(cur_out_device, format, this);
audio_output_->setNotifyInterval(65); // ~15 fps
connect(audio_output_, SIGNAL(stateChanged(QAudio::State)), this, SLOT(outputStateChanged(QAudio::State)));
connect(audio_output_, SIGNAL(notify()), this, SLOT(outputNotify()));
tempfile_->seek(0);
audio_output_->start(tempfile_);
emit startedPlaying();
// QTBUG-6548 StoppedState is not always emitted on error, force a cleanup
// in case playback fails immediately.
if (audio_output_ && audio_output_->state() == QAudio::StoppedState) {
outputStateChanged(QAudio::StoppedState);
}
}
void RtpAudioStream::stopPlaying()
{
if (audio_output_) {
audio_output_->stop();
}
}
void RtpAudioStream::writeSilence(int samples)
{
if (samples < 1 || audio_out_rate_ == 0) return;
unsigned silence_bytes = samples * sample_bytes_;
char *silence_buff = (char *) g_malloc0(silence_bytes);
RTP_STREAM_DEBUG("Writing %u silence samples", samples);
tempfile_->write(silence_buff, silence_bytes);
g_free(silence_buff);
QVector<qint16> visual_fill(samples * visual_sample_rate_ / audio_out_rate_, 0);
visual_samples_ += visual_fill;
}
void RtpAudioStream::outputStateChanged(QAudio::State new_state)
{
if (!audio_output_) return;
// On some platforms including macOS and Windows, the stateChanged signal
// is emitted while a QMutexLocker is active. As a result we shouldn't
// delete audio_output_ here.
switch (new_state) {
case QAudio::StoppedState:
// RTP_STREAM_DEBUG("stopped %f", audio_output_->processedUSecs() / 100000.0);
// Detach from parent (RtpAudioStream) to prevent deleteLater from being
// run during destruction of this class.
audio_output_->setParent(0);
audio_output_->disconnect();
audio_output_->deleteLater();
audio_output_ = NULL;
emit finishedPlaying();
break;
case QAudio::IdleState:
audio_output_->stop();
break;
default:
break;
}
}
void RtpAudioStream::outputNotify()
{
emit processedSecs(audio_output_->processedUSecs() / 1000000.0);
}
#endif // QT_MULTIMEDIA_LIB
/*
* Editor modelines
*
* Local Variables:
* c-basic-offset: 4
* tab-width: 8
* indent-tabs-mode: nil
* End:
*
* ex: set shiftwidth=4 tabstop=8 expandtab:
* :indentSize=4:tabSize=8:noTabs=true:
*/
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