| 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;
|
|
|