webrtc_m130/call/test/mock_audio_send_stream.h

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Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_
#define CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_
#include <memory>
#include "call/audio_send_stream.h"
#include "test/gmock.h"
namespace webrtc {
namespace test {
class MockAudioSendStream : public AudioSendStream {
public:
MOCK_METHOD(const webrtc::AudioSendStream::Config&,
GetConfig,
(),
(const, override));
MOCK_METHOD(void, Reconfigure, (const Config& config), (override));
MOCK_METHOD(void, Start, (), (override));
MOCK_METHOD(void, Stop, (), (override));
Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
// GMock doesn't like move-only types, such as std::unique_ptr.
void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override {
Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
SendAudioDataForMock(audio_frame.get());
}
MOCK_METHOD(void, SendAudioDataForMock, (webrtc::AudioFrame*));
MOCK_METHOD(
bool,
SendTelephoneEvent,
(int payload_type, int payload_frequency, int event, int duration_ms),
(override));
MOCK_METHOD(void, SetMuted, (bool muted), (override));
MOCK_METHOD(Stats, GetStats, (), (const, override));
MOCK_METHOD(Stats, GetStats, (bool has_remote_tracks), (const, override));
Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
};
} // namespace test
} // namespace webrtc
#endif // CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_