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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}
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|---|---|---|
| .. | ||
| test | ||
| audio_receive_stream.h | ||
| audio_send_stream.cc | ||
| audio_send_stream.h | ||
| audio_state.h | ||
| bitrate_allocator.cc | ||
| bitrate_allocator.h | ||
| bitrate_allocator_unittest.cc | ||
| bitrate_estimator_tests.cc | ||
| BUILD.gn | ||
| call.cc | ||
| call.h | ||
| call_perf_tests.cc | ||
| call_unittest.cc | ||
| callfactory.cc | ||
| callfactory.h | ||
| callfactoryinterface.h | ||
| DEPS | ||
| fake_rtp_transport_controller_send.h | ||
| flexfec_receive_stream.h | ||
| flexfec_receive_stream_impl.cc | ||
| flexfec_receive_stream_impl.h | ||
| flexfec_receive_stream_unittest.cc | ||
| OWNERS | ||
| rampup_tests.cc | ||
| rampup_tests.h | ||
| rtcp_demuxer.cc | ||
| rtcp_demuxer.h | ||
| rtcp_demuxer_unittest.cc | ||
| rtcp_packet_sink_interface.h | ||
| rtp_config.cc | ||
| rtp_config.h | ||
| rtp_demuxer.cc | ||
| rtp_demuxer.h | ||
| rtp_demuxer_unittest.cc | ||
| rtp_packet_sink_interface.h | ||
| rtp_rtcp_demuxer_helper.cc | ||
| rtp_rtcp_demuxer_helper.h | ||
| rtp_rtcp_demuxer_helper_unittest.cc | ||
| rtp_stream_receiver_controller.cc | ||
| rtp_stream_receiver_controller.h | ||
| rtp_stream_receiver_controller_interface.h | ||
| rtp_transport_controller_send.cc | ||
| rtp_transport_controller_send.h | ||
| rtp_transport_controller_send_interface.h | ||
| rtx_receive_stream.cc | ||
| rtx_receive_stream.h | ||
| rtx_receive_stream_unittest.cc | ||
| ssrc_binding_observer.h | ||
| syncable.cc | ||
| syncable.h | ||
| video_config.cc | ||
| video_config.h | ||
| video_receive_stream.cc | ||
| video_receive_stream.h | ||
| video_send_stream.cc | ||
| video_send_stream.h | ||