commit 65d61c39248f8c79a1ef28def88fe1f7b06e6a8c
author tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Jun 05 13:42:51 2014 +0000
committer tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Jun 05 13:42:51 2014 +0000
tree ecf1d00c8e98993067c127f65905f03433cd58f6
parent b51d3ea593abd645adba72f59d479564ed01276e

Opus send rate overflows if over 65 kbps

The member holding the send rate for Opus had too low resolution for rates above ~65 kbps.

I've added a test that checks if the average rate in a Opus test is in the right range. The test fails before my fix, and now passes.

BUG=3267
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6344 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/acm2/acm_opus.h

diff --git a/webrtc/modules/audio_coding/main/acm2/acm_opus.h b/webrtc/modules/audio_coding/main/acm2/acm_opus.h
index 7963ee9..07ce072 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_opus.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_opus.h
@@ -47,7 +47,7 @@
 
   WebRtcOpusEncInst* encoder_inst_ptr_;
   uint16_t sample_freq_;
-  uint16_t bitrate_;
+  int32_t bitrate_;
   int channels_;
 
   bool fec_enabled_;

webrtc/modules/audio_coding/main/test/TestStereo.cc

diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/main/test/TestStereo.cc
index f058967..00c3631 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.cc
@@ -807,6 +807,8 @@
   uint32_t time_stamp_diff;
   channel->reset_payload_size();
   int error_count = 0;
+  int variable_bytes = 0;
+  int variable_packets = 0;
 
   while (1) {
     // Simulate packet loss by setting |packet_loss_| to "true" in
@@ -838,11 +840,16 @@
     // Run sender side of ACM
     EXPECT_GT(acm_a_->Process(), -1);
 
-    // Verify that the received packet size matches the settings
+    // Verify that the received packet size matches the settings.
     rec_size = channel->payload_size();
     if ((0 < rec_size) & (rec_size < 65535)) {
-      // Opus is variable rate, skip this test.
-      if (strcmp(send_codec_name_, "opus")) {
+      if (strcmp(send_codec_name_, "opus") == 0) {
+        // Opus is a variable rate codec, hence calculate the average packet
+        // size, and later make sure the average is in the right range.
+        variable_bytes += rec_size;
+        variable_packets++;
+      } else {
+        // For fixed rate codecs, check that packet size is correct.
         if ((rec_size != pack_size_bytes_ * out_channels)
             && (pack_size_bytes_ < 65535)) {
           error_count++;
@@ -866,6 +873,13 @@
 
   EXPECT_EQ(0, error_count);
 
+  // Check that packet size is in the right range for variable rate codecs,
+  // such as Opus.
+  if (variable_packets > 0) {
+    variable_bytes /= variable_packets;
+    EXPECT_NEAR(variable_bytes, pack_size_bytes_, 3);
+  }
+
   if (in_file_mono_->EndOfFile()) {
     in_file_mono_->Rewind();
   }