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Refactored: Moved sample_bytes_ to rtp_media.h as SAMPLE_BYTES

This commit is contained in:
Jirka Novak 2021-02-08 16:47:45 +01:00 committed by Wireshark GitLab Utility
parent 80aa82c039
commit 40c1929115
2 changed files with 21 additions and 21 deletions
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41
ui/qt/rtp_audio_stream.cpp
View File

@ -160,7 +160,6 @@ void RtpAudioStream::reset(double global_start_time, bool stereo, bool left, boo
tempfile_->seek(0);
}
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?
*/
@ -243,7 +242,7 @@ void RtpAudioStream::decode()
QAudioFormat format;
format.setSampleRate(sample_rate);
format.setSampleSize(sample_bytes_ * 8); // bits
format.setSampleSize(SAMPLE_BYTES * 8); // bits
format.setSampleType(QAudioFormat::SignedInt);
if (audio_stereo_) {
format.setChannelCount(2);
@ -292,7 +291,7 @@ void RtpAudioStream::decode()
qint64 silence_samples;
RTP_STREAM_DEBUG("Resync...");
silence_samples = (qint64)((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?
*/
@ -317,7 +316,7 @@ void RtpAudioStream::decode()
if (timing_mode_ == Uninterrupted) {
silence_samples = 0;
} else {
silence_samples = (int)((rtp_time - rtp_time_prev)*sample_rate - decoded_bytes_prev / sample_bytes_);
silence_samples = (int)((rtp_time - rtp_time_prev)*sample_rate - decoded_bytes_prev / SAMPLE_BYTES);
}
if (silence_samples != 0) {
@ -336,7 +335,7 @@ void RtpAudioStream::decode()
// XXX rtp_player.c:696 adds audio here.
rtp_time_prev = rtp_time;
pack_period = (double) decoded_bytes / sample_bytes_ / sample_rate;
pack_period = (double) decoded_bytes / SAMPLE_BYTES / sample_rate;
decoded_bytes_prev = decoded_bytes;
arrive_time_prev = arrive_time;
}
@ -364,31 +363,31 @@ void RtpAudioStream::decode()
}
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))
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);
}
speex_resampler_process_int(audio_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
write_buff = (char *) resample_buff;
write_bytes = out_len * sample_bytes_;
write_bytes = out_len * SAMPLE_BYTES;
}
// Write the decoded, possibly-resampled audio to our temp file.
gint64 silence = 0;
if (audio_stereo_) {
// Process audio mute/left/right settings
for(qint64 i=0; i<write_bytes; i+=sample_bytes_) {
for(qint64 i=0; i<write_bytes; i+=SAMPLE_BYTES) {
if (audio_left_) {
tempfile_->write(write_buff+i, sample_bytes_);
tempfile_->write(write_buff+i, SAMPLE_BYTES);
} else {
tempfile_->write((char *)&silence, sample_bytes_);
tempfile_->write((char *)&silence, SAMPLE_BYTES);
}
if (audio_right_) {
tempfile_->write(write_buff+i, sample_bytes_);
tempfile_->write(write_buff+i, SAMPLE_BYTES);
} else {
tempfile_->write((char *)&silence, sample_bytes_);
tempfile_->write((char *)&silence, SAMPLE_BYTES);
}
}
} else {
@ -396,15 +395,15 @@ void RtpAudioStream::decode()
if (audio_left_) {
tempfile_->write(write_buff, write_bytes);
} else {
writeSilence(write_bytes / sample_bytes_);
writeSilence(write_bytes / SAMPLE_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))
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);
}
@ -601,7 +600,7 @@ void RtpAudioStream::startPlaying()
QAudioFormat format;
format.setSampleRate(audio_out_rate_);
format.setSampleSize(sample_bytes_ * 8); // bits
format.setSampleSize(SAMPLE_BYTES * 8); // bits
format.setSampleType(QAudioFormat::SignedInt);
if (audio_stereo_) {
format.setChannelCount(2);
@ -622,9 +621,9 @@ void RtpAudioStream::startPlaying()
emit playbackError(playback_error);
}
start_pos = (qint64)(start_play_time_ * sample_bytes_ * audio_out_rate_);
// Round to sample_bytes_ boundary
start_pos = (start_pos / sample_bytes_) * sample_bytes_;
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;
@ -680,7 +679,7 @@ void RtpAudioStream::writeSilence(qint64 samples)
{
if (samples < 1 || audio_out_rate_ == 0) return;
qint64 silence_bytes = samples * sample_bytes_;
qint64 silence_bytes = samples * SAMPLE_BYTES;
char *silence_buff = (char *) g_malloc0(silence_bytes);
RTP_STREAM_DEBUG("Writing " G_GUINT64_FORMAT " silence samples", samples);

1
ui/rtp_media.h
View File

@ -33,6 +33,7 @@ extern "C" {
typedef gint16 SAMPLE;
#define SAMPLE_MAX G_MAXINT16
#define SAMPLE_MIN G_MININT16
#define SAMPLE_BYTES (sizeof(SAMPLE) / sizeof(char))
/* Defines an RTP packet */
typedef struct _rtp_packet {