Qt: Use SpeexDSP API names for RTP player

The speex headers use some CPP magic to rename the API.

Using the public names in the calling code reduces obfuscation and
allows linking with libspeexdsp.so also.

Change-Id: I10c6cd2de6b237400224d3db6a9995e646747311
Reviewed-on: https://code.wireshark.org/review/15400
Petri-Dish: João Valverde <j@v6e.pt>
Reviewed-by: Pascal Quantin <pascal.quantin@gmail.com>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: João Valverde <j@v6e.pt>

1 files changed, 14 insertions, 14 deletions

diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 65cb0865e7..8c2ac2d538 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -67,9 +67,9 @@ RtpAudioStream::RtpAudioStream(QObject *parent, _rtp_stream_info *rtp_stream) :
     ssrc_ = rtp_stream->ssrc;
 
     // We keep visual samples in memory. Make fewer of them.
-    visual_resampler_ = ws_codec_resampler_init(1, default_audio_sample_rate_,
+    visual_resampler_ = speex_resampler_init(1, default_audio_sample_rate_,
                                                 visual_sample_rate_, SPEEX_RESAMPLER_QUALITY_MIN, NULL);
-    ws_codec_resampler_skip_zeros(visual_resampler_);
+    speex_resampler_skip_zeros(visual_resampler_);
 
     QString tempname = QString("%1/wireshark_rtp_stream").arg(QDir::tempPath());
     tempfile_ = new QTemporaryFile(tempname, this);
@@ -87,8 +87,8 @@ RtpAudioStream::~RtpAudioStream()
         g_free(rtp_packet);
     }
     g_hash_table_destroy(decoders_hash_);
-    if (audio_resampler_) ws_codec_resampler_destroy (audio_resampler_);
-    ws_codec_resampler_destroy (visual_resampler_);
+    if (audio_resampler_) speex_resampler_destroy (audio_resampler_);
+    speex_resampler_destroy (visual_resampler_);
 }
 
 bool RtpAudioStream::isMatch(const _rtp_stream_info *rtp_stream) const
@@ -159,10 +159,10 @@ void RtpAudioStream::reset(double start_rel_time)
     jitter_drop_timestamps_.clear();
 
     if (audio_resampler_) {
-        ws_codec_resampler_reset_mem(audio_resampler_);
+        speex_resampler_reset_mem(audio_resampler_);
     }
     if (visual_resampler_) {
-        ws_codec_resampler_reset_mem(visual_resampler_);
+        speex_resampler_reset_mem(visual_resampler_);
     }
     tempfile_->seek(0);
 }
@@ -203,7 +203,7 @@ void RtpAudioStream::decode()
         rtp_packet_t *rtp_packet = rtp_packets_[cur_packet];
 
         stop_rel_time_ = start_rel_time_ + rtp_packet->arrive_offset;
-        ws_codec_resampler_get_rate(visual_resampler_, &cur_in_rate, &visual_out_rate);
+        speex_resampler_get_rate(visual_resampler_, &cur_in_rate, &visual_out_rate);
 
         QString payload_name;
         if (rtp_packet->info->info_payload_type_str) {
@@ -324,17 +324,17 @@ void RtpAudioStream::decode()
         if (audio_out_rate_ != sample_rate) {
             // Resample the audio to match our previous output rate.
             if (!audio_resampler_) {
-                audio_resampler_ = ws_codec_resampler_init(1, sample_rate, audio_out_rate_, 10, NULL);
-                ws_codec_resampler_skip_zeros(audio_resampler_);
+                audio_resampler_ = speex_resampler_init(1, sample_rate, audio_out_rate_, 10, NULL);
+                speex_resampler_skip_zeros(audio_resampler_);
                 RTP_STREAM_DEBUG("Started resampling from %u to (out) %u Hz.", sample_rate, audio_out_rate_);
             } else {
                 spx_uint32_t audio_out_rate;
-                ws_codec_resampler_get_rate(audio_resampler_, &cur_in_rate, &audio_out_rate);
+                speex_resampler_get_rate(audio_resampler_, &cur_in_rate, &audio_out_rate);
 
                 // Adjust rates if needed.
                 if (sample_rate != cur_in_rate) {
-                    ws_codec_resampler_set_rate(audio_resampler_, sample_rate, audio_out_rate);
-                    ws_codec_resampler_set_rate(visual_resampler_, sample_rate, visual_out_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_);
                 }
             }
@@ -346,7 +346,7 @@ void RtpAudioStream::decode()
                 resample_buff = (SAMPLE *) g_realloc(resample_buff, resample_buff_len);
             }
 
-            ws_codec_resampler_process_int(audio_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
+            speex_resampler_process_int(audio_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
             write_buff = (char *) decode_buff;
             write_bytes = out_len * sample_bytes_;
         }
@@ -363,7 +363,7 @@ void RtpAudioStream::decode()
             resample_buff = (SAMPLE *) g_realloc(resample_buff, resample_buff_len);
         }
 
-        ws_codec_resampler_process_int(visual_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
+        speex_resampler_process_int(visual_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
         for (unsigned i = 0; i < out_len; i++) {
             packet_timestamps_[stop_rel_time_ + (double) i / visual_out_rate] = rtp_packet->frame_num;
             if (qAbs(resample_buff[i]) > max_sample_val_) max_sample_val_ = qAbs(resample_buff[i]);