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}diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 65e3050..8459385 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -20,8 +20,8 @@
"audio_send_stream.h",
"audio_state.cc",
"audio_state.h",
- "audio_transport_proxy.cc",
- "audio_transport_proxy.h",
+ "audio_transport_impl.cc",
+ "audio_transport_impl.h",
"conversion.h",
"null_audio_poller.cc",
"null_audio_poller.h",
@@ -61,6 +61,8 @@
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../voice_engine",
+ "../voice_engine:audio_level",
+ "utility:audio_frame_operations",
]
}
if (rtc_include_tests) {
@@ -99,6 +101,7 @@
":audio",
":audio_end_to_end_test",
"../api:mock_audio_mixer",
+ "../call:mock_call_interfaces",
"../call:mock_rtp_interfaces",
"../call:rtp_interfaces",
"../call:rtp_receiver",