10#ifndef RTPAUDIOSTREAM_H
11#define RTPAUDIOSTREAM_H
15#ifdef QT_MULTIMEDIA_LIB
31#include <QAudioOutput>
34#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
45class RtpAudioStream :
public QObject
64 explicit RtpAudioStream(QObject *parent,
rtpstream_id_t *
id,
bool stereo_required);
102 void reset(
double global_start_time);
116#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
121 void decode(QAudioDevice out_device);
127 void decode(QAudioDeviceInfo out_device);
139 double startRelTime()
const {
return start_rel_time_; }
145 double stopRelTime()
const {
return stop_rel_time_; }
151 unsigned sampleRate()
const {
return first_sample_rate_; }
157 unsigned playRate()
const {
return audio_out_rate_; }
163 void setRequestedPlayRate(
unsigned new_rate) { audio_requested_out_rate_ = new_rate; }
169 void setVisualSampleRate(
unsigned new_rate) { visual_sample_rate_ = new_rate; }
175 const QStringList payloadNames()
const;
182 const QVector<double> visualTimestamps(
bool relative =
true);
190 const QVector<double> visualSamples(
int y_offset = 0);
197 const QVector<double> outOfSequenceTimestamps(
bool relative =
true);
203 int outOfSequence() {
return static_cast<int>(out_of_seq_timestamps_.size()); }
210 const QVector<double> outOfSequenceSamples(
int y_offset = 0);
217 const QVector<double> jitterDroppedTimestamps(
bool relative =
true);
223 int jitterDropped() {
return static_cast<int>(jitter_drop_timestamps_.size()); }
230 const QVector<double> jitterDroppedSamples(
int y_offset = 0);
237 const QVector<double> wrongTimestampTimestamps(
bool relative =
true);
243 int wrongTimestamps() {
return static_cast<int>(wrong_timestamp_timestamps_.size()); }
250 const QVector<double> wrongTimestampSamples(
int y_offset = 0);
257 const QVector<double> insertedSilenceTimestamps(
bool relative =
true);
263 int insertedSilences() {
return static_cast<int>(silence_timestamps_.size()); }
270 const QVector<double> insertedSilenceSamples(
int y_offset = 0);
278 quint32 nearestPacket(
double timestamp,
bool is_relative =
true);
284 QColor color() {
return color_; }
290 void setColor(QColor color) { color_ = color; }
296 QAudio::State outputState()
const;
302 void setJitterBufferSize(
int jitter_buffer_size) { jitter_buffer_size_ = jitter_buffer_size; }
308 void setTimingMode(TimingMode timing_mode) { timing_mode_ = timing_mode; }
314 void setStartPlayTime(
double start_play_time) { start_play_time_ = start_play_time; }
316#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
322 bool prepareForPlay(QAudioDevice out_device);
329 bool prepareForPlay(QAudioDeviceInfo out_device);
351 void seekPlaying(qint64 samples);
357 void setStereoRequired(
bool stereo_required) { stereo_required_ = stereo_required; }
363 qint16 getMaxSampleValue() {
return max_sample_val_; }
369 void setMaxSampleValue(int16_t max_sample_val) { max_sample_val_ = max_sample_val; }
375 void seekSample(qint64 samples);
382 qint64 readSample(SAMPLE *sample);
388 qint64 getLeadSilenceSamples() {
return prepend_samples_; }
394 qint64 getTotalSamples() {
return (audio_file_->getTotalSamples()); }
400 qint64 getEndOfSilenceSample() {
return (audio_file_->getEndOfSilenceSample()); }
406 double getEndOfSilenceTime() {
return (
double)getEndOfSilenceSample() / (double)playRate(); }
413 qint64 convertTimeToSamples(
double time) {
return (qint64)(time * playRate()); }
420 bool savePayload(QIODevice *file);
426 unsigned getHash() {
return rtpstream_id_to_hash(&(id_)); }
438 QString getIDAsQString();
451 void processedSecs(
double secs);
457 void playbackError(
const QString error_msg);
464 void finishedPlaying(RtpAudioStream *
stream, QAudio::Error error);
475 QVector<struct _rtp_packet *>rtp_packets_;
479 QIODevice *temp_file_;
481 struct _GHashTable *decoders_hash_;
483 double global_start_rel_time_;
485 double start_abs_offset_;
487 double start_rel_time_;
489 double stop_rel_time_;
491 qint64 prepend_samples_;
495 bool stereo_required_;
497 quint32 first_sample_rate_;
499 quint32 audio_out_rate_;
501 quint32 audio_requested_out_rate_;
503 uint32_t visual_sample_rate_;
505 QSet<QString> payload_names_;
507 struct SpeexResamplerState_ *visual_resampler_;
509 QMap<double, quint32> packet_timestamps_;
511 QVector<qint16> visual_samples_;
513 QVector<double> out_of_seq_timestamps_;
515 QVector<double> jitter_drop_timestamps_;
517 QVector<double> wrong_timestamp_timestamps_;
519 QVector<double> silence_timestamps_;
522 qint16 max_sample_val_;
525 qint16 max_sample_val_used_;
531 int jitter_buffer_size_;
534 TimingMode timing_mode_;
537 double start_play_time_;
544 const QString formatDescription(
const QAudioFormat & format);
550 QString currentOutputDevice();
552#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
554 QAudioSink *audio_output_;
560 void decodeAudio(QAudioDevice out_device);
569 quint32 calculateAudioOutRate(QAudioDevice out_device,
unsigned int sample_rate,
unsigned int requested_out_rate);
572 QAudioOutput *audio_output_;
578 void decodeAudio(QAudioDeviceInfo out_device);
587 quint32 calculateAudioOutRate(QAudioDeviceInfo out_device,
unsigned int sample_rate,
unsigned int requested_out_rate);
597 SAMPLE *resizeBufferIfNeeded(SAMPLE *buff, int32_t *buff_bytes, qint64 requested_size);
604 void outputStateChanged(QAudio::State new_state);
609 void delayedStopStream();
Encapsulates the mute state and channel assignment for one audio stream.
Definition rtp_audio_routing.h:35
A QIODevice subclass that handles reading and writing of RTP audio files and frames.
Definition rtp_audio_file.h:49
Represents the metadata and indexing information for a single captured frame.
Definition packet_info.h:43
Definition packet-rtp.h:29
Definition rtp_stream_id.h:33
Holds all state and statistics accumulated for a single RTP stream.
Definition rtp_stream.h:42