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Reducing opus_test runtime to pass Android test

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BUG=2609
R=solenberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5111 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
minyue@webrtc.org
2013-11-11 22:03:52 +00:00
parent e03cafaebc
commit 3e427263ee
1 changed files with 3 additions and 1 deletions
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4
webrtc/modules/audio_coding/main/test/opus_test.cc
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@@ -219,6 +219,7 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
int written_samples = 0; int written_samples = 0;
int read_samples = 0; int read_samples = 0;
int decoded_samples = 0; int decoded_samples = 0;
channel->reset_payload_size(); channel->reset_payload_size();
counter_ = 0; counter_ = 0;
@@ -226,7 +227,8 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate)); EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate)); EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));
while (1) { // Make sure the runtime is less than 60 seconds to pass Android test.
for (size_t audio_length = 0; audio_length < 10000; audio_length += 10) {
bool lost_packet = false; bool lost_packet = false;
// Get 10 msec of audio. // Get 10 msec of audio.