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2a8779763a
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}
45 lines
1.5 KiB
C++
45 lines
1.5 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_
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#define CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_
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#include <memory>
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#include "call/audio_send_stream.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockAudioSendStream : public AudioSendStream {
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public:
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MOCK_CONST_METHOD0(GetConfig, const webrtc::AudioSendStream::Config&());
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MOCK_METHOD1(Reconfigure, void(const Config& config));
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MOCK_METHOD0(Start, void());
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MOCK_METHOD0(Stop, void());
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// GMock doesn't like move-only types, such as std::unique_ptr.
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virtual void SendAudioData(
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std::unique_ptr<webrtc::AudioFrame> audio_frame) {
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SendAudioDataForMock(audio_frame.get());
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}
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MOCK_METHOD1(SendAudioDataForMock,
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void(webrtc::AudioFrame* audio_frame));
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MOCK_METHOD4(SendTelephoneEvent,
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bool(int payload_type, int payload_frequency, int event,
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int duration_ms));
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MOCK_METHOD1(SetMuted, void(bool muted));
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MOCK_CONST_METHOD0(GetStats, Stats());
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MOCK_CONST_METHOD1(GetStats, Stats(bool has_remote_tracks));
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};
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} // namespace test
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} // namespace webrtc
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#endif // CALL_TEST_MOCK_AUDIO_SEND_STREAM_H_
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