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-rw-r--r--epan/dissectors/packet-rtp.h4
-rw-r--r--ui/qt/rtp_audio_stream.cpp242
-rw-r--r--ui/qt/rtp_audio_stream.h7
-rw-r--r--ui/qt/rtp_player_dialog.cpp27
-rw-r--r--ui/qt/rtp_player_dialog.h7
-rw-r--r--ui/qt/rtp_player_dialog.ui2
-rw-r--r--ui/rtp_media.h1
-rw-r--r--ui/rtp_stream.h2
8 files changed, 167 insertions, 125 deletions
diff --git a/epan/dissectors/packet-rtp.h b/epan/dissectors/packet-rtp.h
index d4f381b15b..23aba7cf4d 100644
--- a/epan/dissectors/packet-rtp.h
+++ b/epan/dissectors/packet-rtp.h
@@ -42,8 +42,8 @@ struct _rtp_info {
guint32 info_sync_src;
guint info_data_len; /* length of raw rtp data as reported */
gboolean info_all_data_present; /* FALSE if data is cut off */
- size_t info_payload_offset; /* start of payload relative to info_data */
- size_t info_payload_len; /* length of payload (incl padding) */
+ size_t info_payload_offset; /* start of payload relative to info_data */
+ size_t info_payload_len; /* length of payload (incl padding) */
gboolean info_is_srtp;
guint32 info_setup_frame_num; /* the frame num of the packet that set this RTP connection */
const guint8* info_data; /* pointer to raw rtp data */
diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 68be1ccd4d..0868f5abee 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -19,7 +19,6 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-
#include "rtp_audio_stream.h"
#ifdef QT_MULTIMEDIA_LIB
@@ -40,6 +39,9 @@
#include <QDir>
#include <QTemporaryFile>
+// 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;
@@ -77,6 +79,12 @@ RtpAudioStream::RtpAudioStream(QObject *parent, _rtp_stream_info *rtp_stream) :
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_) ws_codec_resampler_destroy (audio_resampler_);
ws_codec_resampler_destroy (visual_resampler_);
@@ -115,125 +123,27 @@ void RtpAudioStream::addRtpStream(const _rtp_stream_info *rtp_stream)
// RTP_STREAM_DEBUG("added %d:%u packets", g_list_length(rtp_stream->rtp_packet_list), rtp_stream->packet_count);
rtp_streams_ << rtp_stream;
-
- double stream_srt = nstime_to_sec(&rtp_stream->start_rel_time);
- if (rtp_streams_.length() < 2 || stream_srt > start_rel_time_) {
- start_rel_time_ = stop_rel_time_ = stream_srt;
- start_abs_offset_ = nstime_to_sec(&rtp_stream->start_fd->abs_ts) - start_rel_time_;
- }
}
-static const int sample_bytes_ = sizeof(SAMPLE) / sizeof(char);
-void RtpAudioStream::addRtpPacket(const struct _packet_info *pinfo, const _rtp_info *rtp_info)
+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;
- // Combination of gtk/rtp_player.c:decode_rtp_stream + decode_rtp_packet
- // XXX This is more messy than it should be.
-
- SAMPLE *decode_buff = NULL;
- SAMPLE *resample_buff = NULL;
- spx_uint32_t cur_in_rate, visual_out_rate;
- char *write_buff;
- qint64 write_bytes;
- unsigned channels;
- unsigned sample_rate;
- rtp_packet_t rtp_packet;
-
- stop_rel_time_ = nstime_to_sec(&pinfo->rel_ts);
- ws_codec_resampler_get_rate(visual_resampler_, &cur_in_rate, &visual_out_rate);
-
- QString payload_name;
- if (rtp_info->info_payload_type_str) {
- payload_name = rtp_info->info_payload_type_str;
- } else {
- payload_name = try_val_to_str_ext(rtp_info->info_payload_type, &rtp_payload_type_short_vals_ext);
- }
- if (!payload_name.isEmpty()) {
- payload_names_ << payload_name;
- }
-
- // First, decode the payload.
- rtp_packet.info = (_rtp_info *) g_memdup(rtp_info, sizeof(struct _rtp_info));
- rtp_packet.arrive_offset = start_rel_time_;
+ 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_malloc(rtp_info->info_payload_len);
- memcpy(rtp_packet.payload_data, rtp_info->info_data + rtp_info->info_payload_offset, rtp_info->info_payload_len);
- } else {
- rtp_packet.payload_data = NULL;
- }
-
- //size_t decoded_bytes =
- decode_rtp_packet(&rtp_packet, &decode_buff, decoders_hash_, &channels, &sample_rate);
- write_buff = (char *) decode_buff;
- write_bytes = rtp_info->info_payload_len * sample_bytes_;
-
- if (tempfile_->pos() == 0) {
- // First packet. Let it determine our sample rate.
- audio_out_rate_ = sample_rate;
-
- last_sequence_ = rtp_info->info_seq_num - 1;
-
- // Prepend silence to match our sibling streams.
- int prepend_samples = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_;
- if (prepend_samples > 0) {
- int prepend_bytes = prepend_samples * sample_bytes_;
- char *prepend_buff = (char *) g_malloc(prepend_bytes);
- SAMPLE silence = 0;
- memccpy(prepend_buff, &silence, prepend_samples, sample_bytes_);
- tempfile_->write(prepend_buff, prepend_bytes);
- }
- } else if (audio_out_rate_ != sample_rate) {
- // Resample the audio to match our previous output rate.
- if (!audio_resampler_) {
- audio_resampler_ = ws_codec_resampler_init(1, sample_rate, audio_out_rate_, 10, NULL);
- ws_codec_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;
- ws_codec_resampler_get_rate(audio_resampler_, &cur_in_rate, &audio_out_rate);
-
- // Adjust rates if needed.
- if (sample_rate != cur_in_rate) {
- ws_codec_resampler_set_rate(audio_resampler_, sample_rate, audio_out_rate);
- ws_codec_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_info->info_payload_len;
- spx_uint32_t out_len = (audio_out_rate_ * (spx_uint32_t)rtp_info->info_payload_len / sample_rate) + (audio_out_rate_ % sample_rate != 0);
- resample_buff = (SAMPLE *) g_malloc(out_len * sample_bytes_);
-
- ws_codec_resampler_process_int(audio_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
- write_buff = (char *) decode_buff;
- write_bytes = out_len * sample_bytes_;
- }
-
- if (rtp_info->info_seq_num != last_sequence_+1) {
- out_of_seq_timestamps_.append(stop_rel_time_);
- // XXX Add silence to tempfile_ and visual_samples_
+ rtp_packet->payload_data = (guint8 *) g_memdup(&(rtp_info->info_data[rtp_info->info_payload_offset]), rtp_info->info_payload_len);
}
- last_sequence_ = rtp_info->info_seq_num;
- // 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_info->info_payload_len;
- spx_uint32_t out_len = (visual_out_rate * in_len / sample_rate) + (visual_out_rate % sample_rate != 0);
- resample_buff = (SAMPLE *) g_realloc(resample_buff, out_len * sizeof(SAMPLE));
-
- ws_codec_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] = pinfo->fd->num;
- if (qAbs(resample_buff[i]) > max_sample_val_) max_sample_val_ = qAbs(resample_buff[i]);
- visual_samples_.append(resample_buff[i]);
+ if (rtp_packets_.size() < 1) { // First packet
+ start_abs_offset_ = nstime_to_sec(&pinfo->fd->abs_ts) - start_rel_time_;
+ start_rel_time_ = stop_rel_time_ = nstime_to_sec(&pinfo->rel_ts);
}
+ rtp_packet->frame_num = pinfo->fd->num;
+ rtp_packet->arrive_offset = nstime_to_sec(&pinfo->rel_ts) - start_rel_time_;
- // Finally, write the resampled audio to our temp file and clean up.
- g_free(rtp_packet.payload_data);
- g_free(decode_buff);
- g_free(resample_buff);
+ rtp_packets_ << rtp_packet;
}
void RtpAudioStream::reset(double start_rel_time)
@@ -256,6 +166,118 @@ void RtpAudioStream::reset(double start_rel_time)
tempfile_->seek(0);
}
+static const int sample_bytes_ = sizeof(SAMPLE) / sizeof(char);
+void RtpAudioStream::decode()
+{
+ // gtk/rtp_player.c:decode_rtp_stream
+ // XXX This is more messy than it should be.
+
+ SAMPLE *decode_buff = NULL;
+ gsize resample_buff_len = 0x1000;
+ SAMPLE *resample_buff = (SAMPLE *) g_malloc(resample_buff_len);
+ spx_uint32_t cur_in_rate, visual_out_rate;
+ char *write_buff;
+ qint64 write_bytes;
+ unsigned channels;
+ unsigned sample_rate;
+
+ for (int i = 0; i < rtp_packets_.size(); i++) {
+ rtp_packet_t *rtp_packet = rtp_packets_[i];
+
+ stop_rel_time_ = start_rel_time_ + rtp_packet->arrive_offset;
+ ws_codec_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;
+ }
+
+ //size_t decoded_bytes =
+ decode_rtp_packet(rtp_packet, &decode_buff, decoders_hash_, &channels, &sample_rate);
+ write_buff = (char *) decode_buff;
+ write_bytes = rtp_packet->info->info_payload_len * sample_bytes_;
+
+ if (tempfile_->pos() == 0) {
+ // First packet. Let it determine our sample rate.
+ audio_out_rate_ = sample_rate;
+
+ last_sequence_ = rtp_packet->info->info_seq_num - 1;
+
+ // Prepend silence to match our sibling streams.
+ int prepend_samples = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_;
+ if (prepend_samples > 0) {
+ int prepend_bytes = prepend_samples * sample_bytes_;
+ char *prepend_buff = (char *) g_malloc(prepend_bytes);
+ SAMPLE silence = 0;
+ memccpy(prepend_buff, &silence, prepend_samples, sample_bytes_);
+ tempfile_->write(prepend_buff, prepend_bytes);
+ }
+ } else if (audio_out_rate_ != sample_rate) {
+ // Resample the audio to match our previous output rate.
+ if (!audio_resampler_) {
+ audio_resampler_ = ws_codec_resampler_init(1, sample_rate, audio_out_rate_, 10, NULL);
+ ws_codec_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;
+ ws_codec_resampler_get_rate(audio_resampler_, &cur_in_rate, &audio_out_rate);
+
+ // Adjust rates if needed.
+ if (sample_rate != cur_in_rate) {
+ ws_codec_resampler_set_rate(audio_resampler_, sample_rate, audio_out_rate);
+ ws_codec_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);
+ }
+
+ ws_codec_resampler_process_int(audio_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
+ write_buff = (char *) decode_buff;
+ write_bytes = out_len * sample_bytes_;
+ }
+
+ if (rtp_packet->info->info_seq_num != last_sequence_+1) {
+ out_of_seq_timestamps_.append(stop_rel_time_);
+ // XXX Add silence to tempfile_ and visual_samples_
+ }
+ last_sequence_ = rtp_packet->info->info_seq_num;
+
+ // 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);
+ }
+
+ ws_codec_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();
diff --git a/ui/qt/rtp_audio_stream.h b/ui/qt/rtp_audio_stream.h
index 33c3c469ab..f329870161 100644
--- a/ui/qt/rtp_audio_stream.h
+++ b/ui/qt/rtp_audio_stream.h
@@ -40,6 +40,7 @@
class QAudioOutput;
class QTemporaryFile;
+struct _rtp_info;
struct _rtp_stream_info;
struct _rtp_sample;
@@ -54,6 +55,7 @@ public:
void addRtpStream(const struct _rtp_stream_info *rtp_stream);
void addRtpPacket(const struct _packet_info *pinfo, const struct _rtp_info *rtp_info);
void reset(double start_rel_time);
+ void decode();
double startRelTime() const { return start_rel_time_; }
double stopRelTime() const { return stop_rel_time_; }
@@ -103,11 +105,15 @@ public slots:
void stopPlaying();
private:
+ // Used to identify unique streams.
+ // The GTK+ UI also uses the call number + current channel.
address src_addr_;
quint16 src_port_;
address dst_addr_;
quint16 dst_port_;
quint32 ssrc_;
+
+ QVector<struct _rtp_packet *>rtp_packets_;
int last_sequence_;
QTemporaryFile *tempfile_;
struct _GHashTable *decoders_hash_;
@@ -130,7 +136,6 @@ private:
private slots:
void outputStateChanged();
void outputNotify();
-
};
#endif // QT_MULTIMEDIA_LIB
diff --git a/ui/qt/rtp_player_dialog.cpp b/ui/qt/rtp_player_dialog.cpp
index 6a46d4b240..f25668f486 100644
--- a/ui/qt/rtp_player_dialog.cpp
+++ b/ui/qt/rtp_player_dialog.cpp
@@ -177,7 +177,16 @@ RtpPlayerDialog::~RtpPlayerDialog()
delete ui;
}
-void RtpPlayerDialog::retapPackets(bool rescale_axes)
+void RtpPlayerDialog::retapPackets()
+{
+ register_tap_listener("rtp", this, NULL, 0, NULL, tapPacket, NULL);
+ cap_file_.retapPackets();
+ remove_tap_listener(this);
+
+ rescanPackets();
+}
+
+void RtpPlayerDialog::rescanPackets(bool rescale_axes)
{
int row_count = ui->streamTreeWidget->topLevelItemCount();
// Clear existing graphs and reset stream values
@@ -190,10 +199,6 @@ void RtpPlayerDialog::retapPackets(bool rescale_axes)
}
ui->audioPlot->clearGraphs();
- register_tap_listener("rtp", this, NULL, 0, NULL, tapPacket, NULL);
- cap_file_.retapPackets();
- remove_tap_listener(this);
-
bool show_legend = false;
bool relative_timestamps = !ui->todCheckBox->isChecked();
@@ -204,6 +209,8 @@ void RtpPlayerDialog::retapPackets(bool rescale_axes)
RtpAudioStream *audio_stream = ti->data(stream_data_col_, Qt::UserRole).value<RtpAudioStream*>();
int y_offset = row_count - row - 1;
+ audio_stream->decode();
+
// Waveform
QCPGraph *audio_graph = ui->audioPlot->addGraph();
QPen wf_pen(audio_stream->color());
@@ -215,7 +222,7 @@ void RtpPlayerDialog::retapPackets(bool rescale_axes)
audio_graph->setData(audio_stream->visualTimestamps(relative_timestamps), audio_stream->visualSamples(y_offset));
audio_graph->removeFromLegend();
ti->setData(graph_data_col_, Qt::UserRole, QVariant::fromValue<QCPGraph *>(audio_graph));
- // RTP_STREAM_DEBUG("Plotting %s, %d samples", ti->text(src_addr_col_).toUtf8().constData(), audio_graph->data()->keys().length());
+ RTP_STREAM_DEBUG("Plotting %s, %d samples", ti->text(src_addr_col_).toUtf8().constData(), audio_graph->data()->keys().length());
QString span_str = QString("%1 - %2 (%3)")
.arg(QString::number(audio_stream->startRelTime(), 'g', 3))
@@ -303,7 +310,11 @@ void RtpPlayerDialog::addRtpStream(struct _rtp_stream_info *rtp_stream)
} else {
start_rel_time_ = qMin(start_rel_time_, start_rel_time);
}
- // RTP_STREAM_DEBUG("adding stream %s to layout, %u packets, start %u", stream_key.toUtf8().constData(), rtp_stream->packet_count, rtp_stream->start_fd->num);
+ RTP_STREAM_DEBUG("adding stream %d to layout, %u packets, %u in list, start %u",
+ ui->streamTreeWidget->topLevelItemCount(),
+ rtp_stream->packet_count,
+ g_list_length(rtp_stream->rtp_packet_list),
+ rtp_stream->start_fd->num);
}
void RtpPlayerDialog::showEvent(QShowEvent *)
@@ -618,7 +629,7 @@ void RtpPlayerDialog::on_todCheckBox_toggled(bool)
QCPAxis *x_axis = ui->audioPlot->xAxis;
double old_lowest = getLowestTimestamp();
- retapPackets(false);
+ rescanPackets(false);
x_axis->moveRange(getLowestTimestamp() - old_lowest);
ui->audioPlot->replot();
}
diff --git a/ui/qt/rtp_player_dialog.h b/ui/qt/rtp_player_dialog.h
index 4d004b860a..9cb8ed5b9e 100644
--- a/ui/qt/rtp_player_dialog.h
+++ b/ui/qt/rtp_player_dialog.h
@@ -79,9 +79,12 @@ protected:
private slots:
/** Retap the capture file, adding RTP packets that match the
- * streams added using ::addRtpStream and display the dialog.
+ * streams added using ::addRtpStream.
*/
- void retapPackets(bool rescale_axes = true);
+ void retapPackets();
+ /** Clear, decode, and redraw each stream.
+ */
+ void rescanPackets(bool rescale_axes = true);
void updateWidgets();
void graphClicked(QMouseEvent *event);
void mouseMoved(QMouseEvent *);
diff --git a/ui/qt/rtp_player_dialog.ui b/ui/qt/rtp_player_dialog.ui
index 728a67e4db..721f3e5bb5 100644
--- a/ui/qt/rtp_player_dialog.ui
+++ b/ui/qt/rtp_player_dialog.ui
@@ -54,7 +54,7 @@
</column>
<column>
<property name="text">
- <string>First Packet</string>
+ <string>Setup Frame</string>
</property>
</column>
<column>
diff --git a/ui/rtp_media.h b/ui/rtp_media.h
index a5d293cebe..5432091805 100644
--- a/ui/rtp_media.h
+++ b/ui/rtp_media.h
@@ -48,6 +48,7 @@ typedef gint16 SAMPLE;
/* Defines an RTP packet */
typedef struct _rtp_packet {
+ guint32 frame_num; /* Qt only */
struct _rtp_info *info; /* the RTP dissected info */
double arrive_offset; /* arrive offset time since the beginning of the stream in ms */
guint8* payload_data;
diff --git a/ui/rtp_stream.h b/ui/rtp_stream.h
index 2e747ed481..746fb694d0 100644
--- a/ui/rtp_stream.h
+++ b/ui/rtp_stream.h
@@ -117,7 +117,7 @@ struct _rtpstream_tapinfo {
g_free(RTP_STREAM_DEBUG_MSG); \
}
#else
-#define RTP_STREAM_DEBUG()
+#define RTP_STREAM_DEBUG(...)
#endif
/****************************************************************************/
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-30 23:34:33 +0000