diff options
Diffstat (limited to 'ui/qt/rtp_audio_stream.cpp')
| -rw-r--r-- | ui/qt/rtp_audio_stream.cpp | 32 |
1 files changed, 16 insertions, 16 deletions
diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp index 5ceb809f..8950c8df 100644 --- a/ui/qt/rtp_audio_stream.cpp +++ b/ui/qt/rtp_audio_stream.cpp @@ -128,7 +128,7 @@ void RtpAudioStream::addRtpPacket(const struct _packet_info *pinfo, const struct rtp_packet_t *rtp_packet = g_new0(rtp_packet_t, 1); rtp_packet->info = (struct _rtp_info *) g_memdup2(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_memdup2(&(rtp_info->info_data[rtp_info->info_payload_offset]), + rtp_packet->payload_data = (uint8_t *) g_memdup2(&(rtp_info->info_data[rtp_info->info_payload_offset]), rtp_info->info_payload_len); } @@ -258,21 +258,21 @@ void RtpAudioStream::decodeAudio(QAudioDeviceInfo out_device) { // XXX This is more messy than it should be. - gint32 resample_buff_bytes = 0x1000; + int32_t resample_buff_bytes = 0x1000; SAMPLE *resample_buff = (SAMPLE *) g_malloc(resample_buff_bytes); char *write_buff = NULL; qint64 write_bytes = 0; unsigned int channels = 0; unsigned int sample_rate = 0; - guint32 last_sequence = 0; - guint32 last_sequence_w = 0; // Last sequence number we wrote data + uint32_t last_sequence = 0; + uint32_t last_sequence_w = 0; // Last sequence number we wrote data 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; - guint64 start_timestamp = 0; + uint64_t start_timestamp = 0; size_t decoded_bytes_prev = 0; unsigned int audio_resampler_input_rate = 0; @@ -430,7 +430,7 @@ void RtpAudioStream::decodeAudio(QAudioDeviceInfo out_device) // Buffer is in SAMPLEs spx_uint32_t in_len = (spx_uint32_t) (write_bytes / SAMPLE_BYTES); // Output is audio_out_rate_/sample_rate bigger than input - spx_uint32_t out_len = (spx_uint32_t) ((guint64)in_len * audio_out_rate_ / sample_rate); + spx_uint32_t out_len = (spx_uint32_t) ((uint64_t)in_len * audio_out_rate_ / sample_rate); resample_buff = resizeBufferIfNeeded(resample_buff, &resample_buff_bytes, out_len * SAMPLE_BYTES); if (audio_resampler && @@ -472,13 +472,13 @@ void RtpAudioStream::decodeAudio(QAudioDeviceInfo out_device) void RtpAudioStream::decodeVisual() { spx_uint32_t read_len = 0; - gint32 read_buff_bytes = VISUAL_BUFF_BYTES; + int32_t read_buff_bytes = VISUAL_BUFF_BYTES; SAMPLE *read_buff = (SAMPLE *) g_malloc(read_buff_bytes); - gint32 resample_buff_bytes = VISUAL_BUFF_BYTES; + int32_t resample_buff_bytes = VISUAL_BUFF_BYTES; SAMPLE *resample_buff = (SAMPLE *) g_malloc(resample_buff_bytes); unsigned int sample_no = 0; spx_uint32_t out_len; - guint32 frame_num; + uint32_t frame_num; rtp_frame_type type; speex_resampler_set_rate(visual_resampler_, audio_out_rate_, visual_sample_rate_); @@ -486,7 +486,7 @@ void RtpAudioStream::decodeVisual() // Loop over every frame record // readFrameSamples() maintains size of buffer for us while (audio_file_->readFrameSamples(&read_buff_bytes, &read_buff, &read_len, &frame_num, &type)) { - out_len = (spx_uint32_t)(((guint64)read_len * visual_sample_rate_ ) / audio_out_rate_); + out_len = (spx_uint32_t)(((uint64_t)read_len * visual_sample_rate_ ) / audio_out_rate_); if (type == RTP_FRAME_AUDIO) { // We resample only audio samples @@ -538,14 +538,14 @@ const QVector<double> RtpAudioStream::visualTimestamps(bool relative) // Scale the height of the waveform to global scale (max_sample_val_used_) // and adjust its Y offset so that they overlap slightly (stack_offset_). -static const double stack_offset_ = G_MAXINT16 / 3; +static const double stack_offset_ = INT16_MAX / 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++) { if (SAMPLE_NaN != visual_samples_[i]) { - adj_samples.append(((double)visual_samples_[i] * G_MAXINT16 / max_sample_val_used_) + scaled_offset); + adj_samples.append(((double)visual_samples_[i] * INT16_MAX / max_sample_val_used_) + scaled_offset); } else { // Convert to break in graph line adj_samples.append(qQNaN()); @@ -701,8 +701,8 @@ const QString RtpAudioStream::formatDescription(const QAudioFormat &format) QString RtpAudioStream::getIDAsQString() { - gchar *src_addr_str = address_to_display(NULL, &id_.src_addr); - gchar *dst_addr_str = address_to_display(NULL, &id_.dst_addr); + char *src_addr_str = address_to_display(NULL, &id_.src_addr); + char *dst_addr_str = address_to_display(NULL, &id_.dst_addr); QString str = QString("%1:%2 - %3:%4 %5") .arg(src_addr_str) .arg(id_.src_port) @@ -843,7 +843,7 @@ void RtpAudioStream::stopPlaying() if (audio_output_) { if (audio_output_->state() == QAudio::StoppedState) { // Looks like "delayed" QTBUG-6548 - // It may happen that stream is stopped, but no signal emited + // It may happen that stream is stopped, but no signal emitted // Probably triggered by some issue in sound system which is not // handled by Qt correctly outputStateChanged(QAudio::StoppedState); @@ -905,7 +905,7 @@ void RtpAudioStream::delayedStopStream() audio_output_->stop(); } -SAMPLE *RtpAudioStream::resizeBufferIfNeeded(SAMPLE *buff, gint32 *buff_bytes, qint64 requested_size) +SAMPLE *RtpAudioStream::resizeBufferIfNeeded(SAMPLE *buff, int32_t *buff_bytes, qint64 requested_size) { if (requested_size > *buff_bytes) { while ((requested_size > *buff_bytes)) |