Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc |
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc |
index ff6e628696a2bfff658336f1e7c3f9590f64cb0c..11456f21793c34ec53f591c1d0de7ceb74a015bb 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc |
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc |
@@ -17,9 +17,11 @@ |
#include "webrtc/common_types.h" |
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h" |
#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h" |
+#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" |
#include "webrtc/test/field_trial.h" |
#include "webrtc/test/gmock.h" |
#include "webrtc/test/gtest.h" |
+#include "webrtc/test/testsupport/fileutils.h" |
#include "webrtc/system_wrappers/include/clock.h" |
namespace webrtc { |
@@ -113,6 +115,23 @@ void CheckEncoderRuntimeConfig( |
EXPECT_EQ(*config.num_channels, encoder->num_channels_to_encode()); |
} |
+// Create 10ms audio data blocks for a total packet size of "packet_size_ms". |
+std::unique_ptr<test::AudioLoop> Create10msAudioBlocks( |
+ const std::unique_ptr<AudioEncoderOpus>& encoder, |
+ int packet_size_ms) { |
+ const std::string file_name = |
+ test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
+ |
+ std::unique_ptr<test::AudioLoop> speech_data(new test::AudioLoop()); |
+ int audio_samples_per_ms = |
+ rtc::CheckedDivExact(encoder->SampleRateHz(), 1000); |
+ RTC_DCHECK(speech_data->Init( |
hlundin-webrtc
2017/03/03 14:32:59
This is wrong for two reasons:
1. It won't call In
minyue-webrtc
2017/03/03 14:40:41
Right, thanks! Weird that the test still passes.
michaelt
2017/03/06 10:09:52
In general i like the idea of separation of test's
hlundin-webrtc
2017/03/06 10:49:13
The test still passes on the bots (Debug and Relea
michaelt
2017/03/06 12:01:32
Acknowledged.
|
+ file_name, |
+ packet_size_ms * audio_samples_per_ms * encoder->num_channels_to_encode(), |
+ 10 * audio_samples_per_ms * encoder->num_channels_to_encode())); |
+ return speech_data; |
+} |
+ |
} // namespace |
TEST(AudioEncoderOpusTest, DefaultApplicationModeMono) { |
@@ -165,7 +184,7 @@ TEST(AudioEncoderOpusTest, |
OnReceivedUplinkBandwidthWithoutAudioNetworkAdaptor) { |
auto states = CreateCodec(1); |
// Constants are replicated from audio_states.encoderopus.cc. |
- const int kMinBitrateBps = 500; |
+ const int kMinBitrateBps = 6000; |
const int kMaxBitrateBps = 512000; |
// Set a too low bitrate. |
states.encoder->OnReceivedUplinkBandwidth(kMinBitrateBps - 1, |
@@ -183,8 +202,8 @@ TEST(AudioEncoderOpusTest, |
states.encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps, |
rtc::Optional<int64_t>()); |
EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate()); |
- // Set rates from 1000 up to 32000 bps. |
- for (int rate = 1000; rate <= 32000; rate += 1000) { |
+ // Set rates from kMaxBitrateBps up to 32000 bps. |
+ for (int rate = kMinBitrateBps; rate <= 32000; rate += 1000) { |
states.encoder->OnReceivedUplinkBandwidth(rate, rtc::Optional<int64_t>()); |
EXPECT_EQ(rate, states.encoder->GetTargetBitrate()); |
} |
@@ -398,7 +417,7 @@ TEST(AudioEncoderOpusTest, BitrateBounded) { |
test::ScopedFieldTrials override_field_trials( |
"WebRTC-SendSideBwe-WithOverhead/Enabled/"); |
- constexpr int kMinBitrateBps = 500; |
+ constexpr int kMinBitrateBps = 6000; |
constexpr int kMaxBitrateBps = 512000; |
auto states = CreateCodec(2); |
@@ -499,4 +518,31 @@ TEST(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) { |
} |
} |
+TEST(AudioEncoderOpusTest, EncodeAtMinBitrate) { |
+ auto states = CreateCodec(1); |
+ constexpr int kNumPacketsToEncode = 2; |
+ auto audio_frames = |
+ Create10msAudioBlocks(states.encoder, kNumPacketsToEncode * 20); |
+ rtc::Buffer encoded; |
+ uint32_t rtp_timestamp = 12345; // Just a number not important to this test. |
+ |
+ states.encoder->OnReceivedUplinkBandwidth(0, rtc::Optional<int64_t>()); |
+ for (int packet_index = 0; packet_index < kNumPacketsToEncode; |
+ packet_index++) { |
+ // Make sure we are not encoding before we have enough data for |
+ // a 20ms packet. |
+ for (int index = 0; index < 1; index++) { |
+ states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(), |
+ &encoded); |
+ EXPECT_EQ(0u, encoded.size()); |
+ } |
+ |
+ // Should encode now. |
+ states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(), |
+ &encoded); |
+ EXPECT_GT(encoded.size(), 0u); |
+ encoded.Clear(); |
+ } |
+} |
+ |
} // namespace webrtc |