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Unified Diff: webrtc/audio/audio_receive_stream_unittest.cc

Issue 1403363003: Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: missing file Created 5 years, 2 months ago
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Index: webrtc/audio/audio_receive_stream_unittest.cc
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 4e267f17389fa9fbc45e6e122f915fe035c6043c..d3e4bd467d633fd629fdaf3acd7700b96c40aaa0 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -11,15 +11,17 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h"
+#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/test/fake_voice_engine.h"
+namespace webrtc {
+namespace test {
namespace {
-using webrtc::ByteWriter;
-
const size_t kAbsoluteSendTimeLength = 4;
void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
@@ -56,10 +58,31 @@ size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
rtp_header_length += kAbsoluteSendTimeLength;
return rtp_header_length;
}
-} // namespace
-namespace webrtc {
-namespace test {
+struct ConfigHelper {
+ ConfigHelper() {
+ AudioState::Config config;
+ config.voice_engine = &voice_engine_;
+ audio_state_.reset(new internal::AudioState(config));
+ }
+
+ MockRemoteBitrateEstimator* remote_bitrate_estimator() {
+ return &remote_bitrate_estimator_;
+ }
+ AudioReceiveStream::Config& config() {
+ return stream_config_;
+ }
+ internal::AudioState* audio_state() {
+ return audio_state_.get();
+ }
+
+ private:
+ MockRemoteBitrateEstimator remote_bitrate_estimator_;
+ FakeVoiceEngine voice_engine_;
+ rtc::scoped_ptr<internal::AudioState> audio_state_;
+ AudioReceiveStream::Config stream_config_;
+};
+} // namespace
TEST(AudioReceiveStreamTest, ConfigToString) {
const int kAbsSendTimeId = 3;
@@ -77,31 +100,27 @@ TEST(AudioReceiveStreamTest, ConfigToString) {
}
TEST(AudioReceiveStreamTest, ConstructDestruct) {
- MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
- AudioReceiveStream::Config config;
- config.voe_channel_id = 1;
- internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
- &voice_engine);
+ ConfigHelper helper;
+ helper.config().voe_channel_id = 1;
+ internal::AudioReceiveStream recv_stream(
+ helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
}
TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
- MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
- AudioReceiveStream::Config config;
- config.combined_audio_video_bwe = true;
- config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
+ ConfigHelper helper;
+ helper.config().combined_audio_video_bwe = true;
+ helper.config().voe_channel_id = FakeVoiceEngine::kRecvChannelId;
const int kAbsSendTimeId = 3;
- config.rtp.extensions.push_back(
+ helper.config().rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
- &voice_engine);
+ internal::AudioReceiveStream recv_stream(
+ helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
uint8_t rtp_packet[30];
const int kAbsSendTimeValue = 1234;
CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
PacketTime packet_time(5678000, 0);
const size_t kExpectedHeaderLength = 20;
- EXPECT_CALL(remote_bitrate_estimator,
+ EXPECT_CALL(*helper.remote_bitrate_estimator(),
IncomingPacket(packet_time.timestamp / 1000,
sizeof(rtp_packet) - kExpectedHeaderLength, testing::_, false))
.Times(1);
@@ -110,13 +129,11 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
}
TEST(AudioReceiveStreamTest, GetStats) {
- MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
- AudioReceiveStream::Config config;
- config.rtp.remote_ssrc = FakeVoiceEngine::kRecvSsrc;
- config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
- internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
- &voice_engine);
+ ConfigHelper helper;
+ helper.config().rtp.remote_ssrc = FakeVoiceEngine::kRecvSsrc;
+ helper.config().voe_channel_id = FakeVoiceEngine::kRecvChannelId;
+ internal::AudioReceiveStream recv_stream(
+ helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
AudioReceiveStream::Stats stats = recv_stream.GetStats();
const CallStatistics& call_stats = FakeVoiceEngine::kRecvCallStats;

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