Index: webrtc/voice_engine/test/auto_test/voe_conference_test.cc |
diff --git a/webrtc/voice_engine/test/auto_test/voe_conference_test.cc b/webrtc/voice_engine/test/auto_test/voe_conference_test.cc |
index 20a74b46b0880147edae791ffc36161b5be8fae6..d2407f30e4605ed132ea06e1c419e13c0c2acf4e 100644 |
--- a/webrtc/voice_engine/test/auto_test/voe_conference_test.cc |
+++ b/webrtc/voice_engine/test/auto_test/voe_conference_test.cc |
@@ -14,16 +14,28 @@ |
#include "webrtc/base/format_macros.h" |
#include "webrtc/base/timeutils.h" |
#include "webrtc/system_wrappers/interface/sleep.h" |
+#include "webrtc/test/testsupport/fileutils.h" |
#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h" |
namespace { |
- static const int kRttMs = 25; |
+const int kRttMs = 25; |
- static bool IsNear(int ref, int comp, int error) { |
- return (ref - comp <= error) && (comp - ref >= -error); |
+bool IsNear(int ref, int comp, int error) { |
+ return (ref - comp <= error) && (comp - ref >= -error); |
+} |
+ |
+void CreateSilenceFile(const std::string& silence_file, int sample_rate_hz) { |
+ FILE* fid = fopen(silence_file.c_str(), "wb"); |
+ int16_t zero = 0; |
+ for (int i = 0; i < sample_rate_hz; ++i) { |
+ // Write 1 second, but it does not matter since the file will be looped. |
+ fwrite(&zero, sizeof(int16_t), 1, fid); |
} |
+ fclose(fid); |
} |
+} // namespace |
+ |
namespace voetest { |
TEST(VoeConferenceTest, RttAndStartNtpTime) { |
@@ -38,12 +50,16 @@ TEST(VoeConferenceTest, RttAndStartNtpTime) { |
int64_t ntp_delay_; |
}; |
+ const std::string input_file = |
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
+ const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile; |
+ |
const int kDelayMs = 987; |
ConferenceTransport trans; |
trans.SetRtt(kRttMs); |
- unsigned int id_1 = trans.AddStream(); |
- unsigned int id_2 = trans.AddStream(); |
+ unsigned int id_1 = trans.AddStream(input_file, kInputFormat); |
+ unsigned int id_2 = trans.AddStream(input_file, kInputFormat); |
EXPECT_TRUE(trans.StartPlayout(id_1)); |
// Start NTP time is the time when a stream is played out, rather than |
@@ -105,4 +121,56 @@ TEST(VoeConferenceTest, RttAndStartNtpTime) { |
} |
} |
} |
+ |
+ |
+TEST(VoeConferenceTest, ReceivedPackets) { |
+ const int kPackets = 50; |
+ const int kPacketDurationMs = 20; // Correspond to Opus. |
+ |
+ const std::string input_file = |
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
+ const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile; |
+ |
+ const std::string silence_file = |
+ webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence"); |
+ CreateSilenceFile(silence_file, 32000); |
+ |
+ { |
+ ConferenceTransport trans; |
+ // Add silence to stream 0, so that it will be filtered out. |
+ unsigned int id_0 = trans.AddStream(silence_file, kInputFormat); |
+ unsigned int id_1 = trans.AddStream(input_file, kInputFormat); |
+ unsigned int id_2 = trans.AddStream(input_file, kInputFormat); |
+ unsigned int id_3 = trans.AddStream(input_file, kInputFormat); |
+ |
+ EXPECT_TRUE(trans.StartPlayout(id_0)); |
+ EXPECT_TRUE(trans.StartPlayout(id_1)); |
+ EXPECT_TRUE(trans.StartPlayout(id_2)); |
+ EXPECT_TRUE(trans.StartPlayout(id_3)); |
+ |
+ webrtc::SleepMs(kPacketDurationMs * kPackets); |
+ |
+ webrtc::CallStatistics stats_0; |
+ webrtc::CallStatistics stats_1; |
+ webrtc::CallStatistics stats_2; |
+ webrtc::CallStatistics stats_3; |
+ EXPECT_TRUE(trans.GetReceiverStatistics(id_0, &stats_0)); |
+ EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1)); |
+ EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2)); |
+ EXPECT_TRUE(trans.GetReceiverStatistics(id_3, &stats_3)); |
+ |
+ // We expect stream 0 to be filtered out totally, but since it may join the |
+ // call earlier than other streams and the beginning packets might have got |
+ // through. So we only expect |packetsReceived| to be close to zero. |
+ EXPECT_NEAR(stats_0.packetsReceived, 0, 2); |
+ // We expect |packetsReceived| to match |kPackets|, but the actual value |
+ // depends on the sleep timer. So we allow a small off from |kPackets|. |
+ EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2); |
+ EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2); |
+ EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2); |
+ } |
+ |
+ remove(silence_file.c_str()); |
+} |
+ |
} // namespace voetest |