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..caadae8cbdc97659c5c73dfeab42dc733cc52dac 100644 |
--- a/webrtc/voice_engine/test/auto_test/voe_conference_test.cc |
+++ b/webrtc/voice_engine/test/auto_test/voe_conference_test.cc |
@@ -14,6 +14,7 @@ |
#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 { |
@@ -27,6 +28,10 @@ namespace { |
namespace voetest { |
TEST(VoeConferenceTest, RttAndStartNtpTime) { |
+ const std::string kInputFileName = |
Andrew MacDonald
2015/08/05 16:37:07
nit: not a compile-time const.
minyue-webrtc
2015/08/06 13:31:22
Done.
minyue-webrtc
2015/08/06 14:57:14
Hi Andrew,
Per offline talk with Tina, I think I
Andrew MacDonald
2015/08/13 19:48:11
Right, it's not a compile-time const (i.e. can onl
|
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
+ const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile; |
+ |
struct Stats { |
Stats(int64_t rtt_receiver_1, int64_t rtt_receiver_2, int64_t ntp_delay) |
: rtt_receiver_1_(rtt_receiver_1), |
@@ -42,8 +47,8 @@ TEST(VoeConferenceTest, RttAndStartNtpTime) { |
ConferenceTransport trans; |
trans.SetRtt(kRttMs); |
- unsigned int id_1 = trans.AddStream(); |
- unsigned int id_2 = trans.AddStream(); |
+ unsigned int id_1 = trans.AddStream(kInputFileName, kInputFormat); |
+ unsigned int id_2 = trans.AddStream(kInputFileName, kInputFormat); |
EXPECT_TRUE(trans.StartPlayout(id_1)); |
// Start NTP time is the time when a stream is played out, rather than |
@@ -105,4 +110,46 @@ TEST(VoeConferenceTest, RttAndStartNtpTime) { |
} |
} |
} |
+ |
+ |
+TEST(VoeConferenceTest, ReceivedPackets) { |
+ const std::string kInputFileName = |
+ webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
+ const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile; |
+ const int kPackets = 50; |
+ const int kPacketDurationMs = 20; // Correspond to Opus. |
+ |
+ ConferenceTransport trans; |
+ // Add silence to stream 0, so that it will be filtered out. |
+ unsigned int id_0 = trans.AddStream( |
+ webrtc::test::ResourcePath("audio_coding/silence", "pcm"), |
+ kInputFormat); |
+ unsigned int id_1 = trans.AddStream(kInputFileName, kInputFormat); |
+ unsigned int id_2 = trans.AddStream(kInputFileName, kInputFormat); |
+ unsigned int id_3 = trans.AddStream(kInputFileName, 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)); |
+ |
+ // Cannot be accurate since stream 0 started the earliest. |
tlegrand-webrtc
2015/08/05 13:32:44
Can you explain this comment? I'm not following.
minyue-webrtc
2015/08/06 13:31:22
I have updated the comments.
tlegrand-webrtc
2015/08/06 14:51:55
Acknowledged.
|
+ EXPECT_NEAR(stats_0.packetsReceived, 0, 2); |
+ // Cannot be accurate since it replies on the sleep timer. |
+ EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2); |
+ EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2); |
+ EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2); |
+} |
+ |
} // namespace voetest |