| Index: webrtc/audio/test/audio_end_to_end_test.h
|
| diff --git a/webrtc/audio/test/low_bandwidth_audio_test.h b/webrtc/audio/test/audio_end_to_end_test.h
|
| similarity index 64%
|
| rename from webrtc/audio/test/low_bandwidth_audio_test.h
|
| rename to webrtc/audio/test/audio_end_to_end_test.h
|
| index ae75707f66d3b64d3f7b1d707ec8dab8d2b34db1..d14b7a108f6785541f17cb61907568ca4a7022dc 100644
|
| --- a/webrtc/audio/test/low_bandwidth_audio_test.h
|
| +++ b/webrtc/audio/test/audio_end_to_end_test.h
|
| @@ -7,28 +7,28 @@
|
| * in the file PATENTS. All contributing project authors may
|
| * be found in the AUTHORS file in the root of the source tree.
|
| */
|
| -#ifndef WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
|
| -#define WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
|
| +#ifndef WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
|
| +#define WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
|
|
|
| #include <memory>
|
| #include <string>
|
| #include <vector>
|
|
|
| #include "webrtc/test/call_test.h"
|
| -#include "webrtc/test/fake_audio_device.h"
|
|
|
| namespace webrtc {
|
| namespace test {
|
|
|
| -class AudioQualityTest : public test::EndToEndTest {
|
| +class AudioEndToEndTest : public test::EndToEndTest {
|
| public:
|
| - AudioQualityTest();
|
| + AudioEndToEndTest();
|
|
|
| protected:
|
| - virtual std::string AudioInputFile();
|
| - virtual std::string AudioOutputFile();
|
| + test::FakeAudioDevice* send_audio_device() { return send_audio_device_; }
|
| + const AudioSendStream* send_stream() const { return send_stream_; }
|
| + const AudioReceiveStream* receive_stream() const { return receive_stream_; }
|
|
|
| - virtual FakeNetworkPipe::Config GetNetworkPipeConfig();
|
| + virtual FakeNetworkPipe::Config GetNetworkPipeConfig() const;
|
|
|
| size_t GetNumVideoStreams() const override;
|
| size_t GetNumAudioStreams() const override;
|
| @@ -50,15 +50,19 @@ class AudioQualityTest : public test::EndToEndTest {
|
| void ModifyAudioConfigs(
|
| AudioSendStream::Config* send_config,
|
| std::vector<AudioReceiveStream::Config>* receive_configs) override;
|
| + void OnAudioStreamsCreated(
|
| + AudioSendStream* send_stream,
|
| + const std::vector<AudioReceiveStream*>& receive_streams) override;
|
|
|
| void PerformTest() override;
|
| - void OnTestFinished() override;
|
|
|
| private:
|
| - test::FakeAudioDevice* send_audio_device_;
|
| + test::FakeAudioDevice* send_audio_device_ = nullptr;
|
| + AudioSendStream* send_stream_ = nullptr;
|
| + AudioReceiveStream* receive_stream_ = nullptr;
|
| };
|
|
|
| } // namespace test
|
| } // namespace webrtc
|
|
|
| -#endif // WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
|
| +#endif // WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
|
|
|