diff options
Diffstat (limited to 'ui/qt/rtp_audio_stream.cpp')
| -rw-r--r-- | ui/qt/rtp_audio_stream.cpp | 69 |
1 files changed, 56 insertions, 13 deletions
diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp index 443f18d1af..b729ab4f62 100644 --- a/ui/qt/rtp_audio_stream.cpp +++ b/ui/qt/rtp_audio_stream.cpp @@ -133,10 +133,13 @@ void RtpAudioStream::addRtpPacket(const struct _packet_info *pinfo, const struct rtp_packets_ << rtp_packet; } -void RtpAudioStream::reset(double global_start_time) +void RtpAudioStream::reset(double global_start_time, bool stereo, bool left, bool right) { global_start_rel_time_ = global_start_time; stop_rel_time_ = start_rel_time_; + audio_stereo_ = stereo; + audio_left_ = left; + audio_right_ = right; audio_out_rate_ = 0; max_sample_val_ = 1; packet_timestamps_.clear(); @@ -157,7 +160,7 @@ 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; +static const qint64 max_silence_samples_ = MAX_SILENCE_FRAMES; void RtpAudioStream::decode() { @@ -238,7 +241,11 @@ void RtpAudioStream::decode() format.setSampleRate(sample_rate); format.setSampleSize(sample_bytes_ * 8); // bits format.setSampleType(QAudioFormat::SignedInt); - format.setChannelCount(1); + if (audio_stereo_) { + format.setChannelCount(2); + } else { + format.setChannelCount(1); + } format.setCodec("audio/pcm"); if (!cur_out_device.isFormatSupported(format)) { @@ -250,7 +257,7 @@ void RtpAudioStream::decode() // Prepend silence to match our sibling streams. tempfile_->seek(0); - int prepend_samples = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_; + qint64 prepend_samples = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_; if (prepend_samples > 0) { writeSilence(prepend_samples); } @@ -278,10 +285,10 @@ void RtpAudioStream::decode() /* 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; + qint64 silence_samples; RTP_STREAM_DEBUG("Resync..."); - silence_samples = (int)((arrive_time - arrive_time_prev)*sample_rate - decoded_bytes_prev / sample_bytes_); + silence_samples = (qint64)((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? */ @@ -301,7 +308,7 @@ void RtpAudioStream::decode() } else { // rtp_player.c:664 /* Add silence if it is necessary */ - int silence_samples; + qint64 silence_samples; if (timing_mode_ == Uninterrupted) { silence_samples = 0; @@ -365,7 +372,29 @@ void RtpAudioStream::decode() } // Write the decoded, possibly-resampled audio to our temp file. - tempfile_->write(write_buff, write_bytes); + gint64 silence = 0; + if (audio_stereo_) { + // Process audio mute/left/right settings + for(qint64 i=0; i<write_bytes; i+=sample_bytes_) { + if (audio_left_) { + tempfile_->write(write_buff+i, sample_bytes_); + } else { + tempfile_->write((char *)&silence, sample_bytes_); + } + if (audio_right_) { + tempfile_->write(write_buff+i, sample_bytes_); + } else { + tempfile_->write((char *)&silence, sample_bytes_); + } + } + } else { + // Process audio mute/unmute settings + if (audio_left_) { + tempfile_->write(write_buff, write_bytes); + } else { + writeSilence(write_bytes / sample_bytes_); + } + } // Collect our visual samples. spx_uint32_t in_len = (spx_uint32_t)rtp_packet->info->info_payload_len; @@ -570,7 +599,11 @@ void RtpAudioStream::startPlaying() format.setSampleRate(audio_out_rate_); format.setSampleSize(sample_bytes_ * 8); // bits format.setSampleType(QAudioFormat::SignedInt); - format.setChannelCount(1); + if (audio_stereo_) { + format.setChannelCount(2); + } else { + format.setChannelCount(1); + } format.setCodec("audio/pcm"); // RTP_STREAM_DEBUG("playing %s %d samples @ %u Hz", @@ -592,6 +625,10 @@ void RtpAudioStream::startPlaying() start_pos = (qint64)(start_play_time_ * sample_bytes_ * audio_out_rate_); // Round to sample_bytes_ boundary start_pos = (start_pos / sample_bytes_) * sample_bytes_; + if (audio_stereo_) { + // There is 2x more samples for stereo + start_pos *= 2; + } if (start_pos < tempfile_->size()) { tempfile_->seek(start_pos); audio_output_->start(tempfile_); @@ -614,15 +651,21 @@ void RtpAudioStream::stopPlaying() } } -void RtpAudioStream::writeSilence(int samples) +void RtpAudioStream::writeSilence(qint64 samples) { if (samples < 1 || audio_out_rate_ == 0) return; - unsigned silence_bytes = samples * sample_bytes_; + qint64 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); + RTP_STREAM_DEBUG("Writing %llu silence samples", samples); + if (audio_stereo_) { + // Silence for left and right channel + tempfile_->write(silence_buff, silence_bytes); + tempfile_->write(silence_buff, silence_bytes); + } else { + tempfile_->write(silence_buff, silence_bytes); + } g_free(silence_buff); // Silence is inserted to audio file only. |