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-rw-r--r--ui/qt/rtp_audio_stream.cpp142
-rw-r--r--ui/qt/rtp_audio_stream.h15
2 files changed, 114 insertions, 43 deletions
diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 091992c14a..44fe3b8f42 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -65,8 +65,10 @@ RtpAudioStream::RtpAudioStream(QObject *parent, rtpstream_info_t *rtpstream, boo
speex_resampler_skip_zeros(visual_resampler_);
QString tempname = QString("%1/wireshark_rtp_stream").arg(QDir::tempPath());
- samplefile_ = new QTemporaryFile(tempname, this);
- samplefile_->open();
+ sample_file_ = new QTemporaryFile(tempname, this);
+ sample_file_->open();
+ sample_file_frame_ = new QTemporaryFile(tempname, this);
+ sample_file_frame_->open();
// RTP_STREAM_DEBUG("Writing to %s", tempname.toUtf8().constData());
}
@@ -83,8 +85,9 @@ RtpAudioStream::~RtpAudioStream()
if (audio_resampler_) speex_resampler_destroy (audio_resampler_);
speex_resampler_destroy (visual_resampler_);
rtpstream_id_free(&id_);
- if (samplefile_) delete samplefile_;
- // tempfile_ is released by audio_output_
+ if (sample_file_) delete sample_file_;
+ if (sample_file_frame_) delete sample_file_frame_;
+ // temp_file_ is released by audio_output_
if (audio_output_) delete audio_output_;
}
@@ -154,10 +157,9 @@ void RtpAudioStream::reset(double global_start_time)
if (audio_resampler_) {
speex_resampler_reset_mem(audio_resampler_);
}
- if (visual_resampler_) {
- speex_resampler_reset_mem(visual_resampler_);
- }
- samplefile_->seek(0);
+ speex_resampler_reset_mem(visual_resampler_);
+ sample_file_->seek(0);
+ sample_file_frame_->seek(0);
}
AudioRouting RtpAudioStream::getAudioRouting()
@@ -179,12 +181,20 @@ void RtpAudioStream::decode(QAudioDeviceInfo out_device)
{
if (rtp_packets_.size() < 1) return;
+ decodeAudio(out_device);
+
+ speex_resampler_reset_mem(visual_resampler_);
+ decodeVisual();
+}
+
+void RtpAudioStream::decodeAudio(QAudioDeviceInfo out_device)
+{
// 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;
+ spx_uint32_t cur_in_rate = 0;
char *write_buff = NULL;
qint64 write_bytes = 0;
unsigned channels = 0;
@@ -200,13 +210,14 @@ void RtpAudioStream::decode(QAudioDeviceInfo out_device)
size_t decoded_bytes_prev = 0;
+ _rtp_packet_frame frame_info;
+
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) {
@@ -261,10 +272,10 @@ void RtpAudioStream::decode(QAudioDeviceInfo out_device)
RTP_STREAM_DEBUG("Audio sample rate is %u", audio_out_rate_);
// Prepend silence to match our sibling streams.
- samplefile_->seek(0);
- qint64 prepend_samples = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_;
- if (prepend_samples > 0) {
- writeSilence(prepend_samples);
+ sample_file_->seek(0);
+ prepend_samples_ = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_;
+ if (prepend_samples_ > 0) {
+ writeSilence(prepend_samples_);
}
}
@@ -302,8 +313,8 @@ void RtpAudioStream::decode(QAudioDeviceInfo out_device)
silence_timestamps_.append(stop_rel_time_);
decoded_bytes_prev = 0;
-/* defined start_timestamp 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") */
+ /* defined start_timestamp 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;
@@ -342,7 +353,7 @@ void RtpAudioStream::decode(QAudioDeviceInfo out_device)
arrive_time_prev = arrive_time;
}
- // Write samples to our file.
+ // Prepare samples to write
write_buff = (char *) decode_buff;
write_bytes = decoded_bytes;
@@ -359,7 +370,6 @@ void RtpAudioStream::decode(QAudioDeviceInfo out_device)
// 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_);
}
}
@@ -377,28 +387,78 @@ void RtpAudioStream::decode(QAudioDeviceInfo out_device)
}
// Write the decoded, possibly-resampled audio to our temp file.
- samplefile_->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);
+ sample_file_->write(write_buff, write_bytes);
+
+ // Record frame info to separate file
+ frame_info.len = write_bytes;
+ frame_info.frame_num = rtp_packet->frame_num;
+ sample_file_frame_->write((char *)&frame_info, sizeof(frame_info));
+
+ // Finally, write the resampled audio to our temp file and clean up.
+ g_free(decode_buff);
+ }
+ g_free(resample_buff);
+}
+
+// We preallocate buffer, 320 samples is enough for most scenarios
+#define VISUAL_BUFF_LEN (320)
+#define VISUAL_BUFF_BYTES (SAMPLE_BYTES * VISUAL_BUFF_LEN)
+void RtpAudioStream::decodeVisual()
+{
+ spx_uint32_t cur_in_rate = 0, visual_out_rate = 0;
+ guint64 read_bytes = 0;
+ spx_uint32_t read_len = 0;
+ gint32 read_buff_bytes = VISUAL_BUFF_BYTES;
+ SAMPLE *read_buff = (SAMPLE *) g_malloc(read_buff_bytes);
+ gint32 resample_buff_bytes = VISUAL_BUFF_BYTES;
+ SAMPLE *resample_buff = (SAMPLE *) g_malloc(resample_buff_bytes);
+ unsigned int sample_no = 0;
+ _rtp_packet_frame frame_info;
+ spx_uint32_t out_len;
+
+ speex_resampler_get_rate(visual_resampler_, &cur_in_rate, &visual_out_rate);
+
+ // Skip silence at begin of the stream
+ sample_file_->seek(prepend_samples_ * SAMPLE_BYTES);
+ sample_file_frame_->seek(0);
+
+ // Loop over every frame record
+ while(sample_file_frame_->read((char *)&frame_info, sizeof(frame_info))) {
+ // Resize buffer when needed
+ if (frame_info.len > read_buff_bytes) {
+ while ((frame_info.len > read_buff_bytes)) {
+ read_buff_bytes *= 2;
+ }
+ read_buff = (SAMPLE *) g_realloc(read_buff, read_buff_bytes);
}
- speex_resampler_process_int(visual_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
+ read_bytes = sample_file_->read((char *)read_buff, frame_info.len);
+ read_len = (spx_uint32_t)read_bytes / SAMPLE_BYTES;
+ out_len = (visual_out_rate * read_len / audio_out_rate_) + (visual_out_rate % audio_out_rate_ != 0);
+
+ // Resize buffer when needed
+ if (out_len * SAMPLE_BYTES > (spx_uint32_t)resample_buff_bytes) {
+ while ((out_len * SAMPLE_BYTES > (spx_uint32_t)resample_buff_bytes)) {
+ resample_buff_bytes *= 2;
+ }
+ resample_buff = (SAMPLE *) g_realloc(resample_buff, resample_buff_bytes);
+ }
+
+ // Resample
+ speex_resampler_process_int(visual_resampler_, 0, read_buff, &read_len, resample_buff, &out_len);
+
+ // Create timestamp and visual sample
for (unsigned i = 0; i < out_len; i++) {
- packet_timestamps_[stop_rel_time_ + (double) i / visual_out_rate] = rtp_packet->frame_num;
+ double time = start_rel_time_ + (double) sample_no / visual_out_rate;
+ packet_timestamps_[time] = frame_info.frame_num;
if (qAbs(resample_buff[i]) > max_sample_val_) max_sample_val_ = qAbs(resample_buff[i]);
visual_samples_.append(resample_buff[i]);
+ sample_no++;
}
-
- // Finally, write the resampled audio to our temp file and clean up.
- g_free(decode_buff);
}
+
g_free(resample_buff);
+ g_free(read_buff);
}
const QStringList RtpAudioStream::payloadNames() const
@@ -587,8 +647,8 @@ bool RtpAudioStream::prepareForPlay(QAudioDeviceInfo out_device)
format.setCodec("audio/pcm");
// RTP_STREAM_DEBUG("playing %s %d samples @ %u Hz",
- // samplefile_->fileName().toUtf8().constData(),
- // (int) samplefile_->size(), audio_out_rate_);
+ // sample_file_->fileName().toUtf8().constData(),
+ // (int) sample_file_->size(), audio_out_rate_);
if (!out_device.isFormatSupported(format)) {
QString playback_error = tr("%1 does not support PCM at %2. Preferred format is %3")
@@ -601,7 +661,7 @@ bool RtpAudioStream::prepareForPlay(QAudioDeviceInfo out_device)
start_pos = (qint64)(start_play_time_ * SAMPLE_BYTES * audio_out_rate_);
// Round to SAMPLE_BYTES boundary
start_pos = (start_pos / SAMPLE_BYTES) * SAMPLE_BYTES;
- size = samplefile_->size();
+ size = sample_file_->size();
if (stereo_required_) {
// There is 2x more samples for stereo
start_pos *= 2;
@@ -611,15 +671,15 @@ bool RtpAudioStream::prepareForPlay(QAudioDeviceInfo out_device)
int buffer_size;
// Start and stop audio with no connection to UI and store buffer size
- tempfile_ = new AudioRoutingFilter(samplefile_, stereo_required_, audio_routing_);
- tempfile_->seek(start_pos);
+ temp_file_ = new AudioRoutingFilter(sample_file_, stereo_required_, audio_routing_);
+ temp_file_->seek(start_pos);
audio_output_ = new QAudioOutput(out_device, format, this);
- audio_output_->start(tempfile_);
+ audio_output_->start(temp_file_);
buffer_size = audio_output_->bufferSize();
audio_output_->stop();
// Start audio again with trippled buffer size
- tempfile_->seek(start_pos);
+ temp_file_->seek(start_pos);
audio_output_->setBufferSize(buffer_size*3);
audio_output_->setNotifyInterval(100); // ~15 fps
connect(audio_output_, SIGNAL(stateChanged(QAudio::State)), this, SLOT(outputStateChanged(QAudio::State)));
@@ -635,7 +695,7 @@ bool RtpAudioStream::prepareForPlay(QAudioDeviceInfo out_device)
void RtpAudioStream::startPlaying()
{
- audio_output_->start(tempfile_);
+ audio_output_->start(temp_file_);
// 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) {
@@ -675,7 +735,7 @@ void RtpAudioStream::writeSilence(qint64 samples)
char *silence_buff = (char *) g_malloc0(silence_bytes);
RTP_STREAM_DEBUG("Writing " G_GUINT64_FORMAT " silence samples", samples);
- samplefile_->write(silence_buff, silence_bytes);
+ sample_file_->write(silence_buff, silence_bytes);
g_free(silence_buff);
// Silence is inserted to audio file only.
diff --git a/ui/qt/rtp_audio_stream.h b/ui/qt/rtp_audio_stream.h
index 9e3369a7bd..78eafbdbc1 100644
--- a/ui/qt/rtp_audio_stream.h
+++ b/ui/qt/rtp_audio_stream.h
@@ -36,6 +36,12 @@ class QTemporaryFile;
struct _rtp_info;
struct _rtp_sample;
+// Structure used for storing frame num during visual waveform decoding
+typedef struct {
+ qint64 len;
+ guint32 frame_num;
+} _rtp_packet_frame;
+
class RtpAudioStream : public QObject
{
Q_OBJECT
@@ -150,8 +156,9 @@ private:
rtpstream_id_t id_;
QVector<struct _rtp_packet *>rtp_packets_;
- QTemporaryFile *samplefile_;
- QIODevice *tempfile_;
+ QTemporaryFile *sample_file_; // Stores waveform samples
+ QTemporaryFile *sample_file_frame_; // Stores _rtp_packet_frame per packet
+ QIODevice *temp_file_;
struct _GHashTable *decoders_hash_;
// TODO: It is not used
//QList<const rtpstream_info_t *>rtpstreams_;
@@ -159,6 +166,7 @@ private:
double start_abs_offset_;
double start_rel_time_;
double stop_rel_time_;
+ qint64 prepend_samples_; // Count of silence samples to match other streams
AudioRouting audio_routing_;
bool stereo_required_;
quint32 audio_out_rate_;
@@ -183,6 +191,9 @@ private:
const QString formatDescription(const QAudioFormat & format);
QString currentOutputDevice();
+ void decodeAudio(QAudioDeviceInfo out_device);
+ void decodeVisual();
+
private slots:
void outputStateChanged(QAudio::State new_state);
};
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-31 02:19:00 +0000