2016-03-17 12:05:36 +01:00
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|
include_rules = [
|
2017-09-15 06:47:31 +02:00
|
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|
"+call",
|
|
|
|
|
"+common_audio",
|
|
|
|
|
"+logging/rtc_event_log",
|
|
|
|
|
"+modules/audio_coding",
|
|
|
|
|
"+modules/audio_device",
|
|
|
|
|
"+modules/audio_mixer",
|
2018-01-17 11:18:31 +01:00
|
|
|
"+modules/audio_processing",
|
2017-09-15 06:47:31 +02:00
|
|
|
"+modules/audio_processing/include",
|
|
|
|
|
"+modules/bitrate_controller",
|
|
|
|
|
"+modules/congestion_controller",
|
|
|
|
|
"+modules/pacing",
|
|
|
|
|
"+modules/remote_bitrate_estimator",
|
|
|
|
|
"+modules/rtp_rtcp",
|
2018-01-17 11:18:31 +01:00
|
|
|
"+modules/utility",
|
2017-09-15 06:47:31 +02:00
|
|
|
"+system_wrappers",
|
2016-03-17 12:05:36 +01:00
|
|
|
]
|
2017-04-27 02:08:52 -07:00
|
|
|
|
|
|
|
|
specific_include_rules = {
|
|
|
|
|
"audio_send_stream.cc": [
|
2017-09-15 06:47:31 +02:00
|
|
|
"+modules/audio_coding/codecs/cng/audio_encoder_cng.h",
|
2017-04-27 02:08:52 -07:00
|
|
|
],
|
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
|
|
|
"audio_transport_impl.h": [
|
|
|
|
|
"+modules/audio_processing/typing_detection.h",
|
|
|
|
|
]
|
2017-04-27 02:08:52 -07:00
|
|
|
}
|