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Revert "Activate ACM test for Android in modules_tests." (rev5364).

Browse Source 
TBR=turaj@webrtc.org,tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5372 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andresp@webrtc.org
2014-01-13 13:15:59 +00:00
parent 44461fa5cb
commit d0b436a935
5 changed files with 48 additions and 293 deletions
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20
webrtc/modules/audio_coding/main/test/TestStereo.cc
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@ -809,14 +809,7 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
channel->reset_payload_size();
int error_count = 0;
#ifdef WEBRTC_ARCH_ARM
const int kMaxNumProcessedFrames = 100; // Limit to 1 second of audio.
#else
const int kMaxNumProcessedFrames = 3000; // Limit to 30 second of audio.
#endif
int num_frames = 0;
while (num_frames < kMaxNumProcessedFrames) {
while (1) {
// Simulate packet loss by setting |packet_loss_| to "true" in
// |percent_loss| percent of the loops.
if (percent_loss > 0) {
@ -870,15 +863,16 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
out_file_.Write10MsData(
audio_frame.data_,
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
++num_frames;
}
EXPECT_EQ(0, error_count);
in_file_mono_->Rewind();
in_file_stereo_->Rewind();
if (in_file_mono_->EndOfFile()) {
in_file_mono_->Rewind();
}
if (in_file_stereo_->EndOfFile()) {
in_file_stereo_->Rewind();
}
// Reset in case we ended with a lost packet
channel->set_lost_packet(false);
}