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/audio_state.h b/audio/audio_state.h
index f4bddbf..14dc788 100644
--- a/audio/audio_state.h
+++ b/audio/audio_state.h
@@ -11,9 +11,10 @@
#ifndef AUDIO_AUDIO_STATE_H_
#define AUDIO_AUDIO_STATE_H_
+#include <map>
#include <memory>
-#include "audio/audio_transport_proxy.h"
+#include "audio/audio_transport_impl.h"
#include "audio/null_audio_poller.h"
#include "audio/scoped_voe_interface.h"
#include "call/audio_state.h"
@@ -24,6 +25,9 @@
#include "voice_engine/include/voe_base.h"
namespace webrtc {
+
+class AudioSendStream;
+
namespace internal {
class AudioState final : public webrtc::AudioState {
@@ -36,21 +40,30 @@
return config_.audio_processing.get();
}
AudioTransport* audio_transport() override {
- return &audio_transport_proxy_;
+ return &audio_transport_;
}
void SetPlayout(bool enabled) override;
void SetRecording(bool enabled) override;
+ Stats GetAudioInputStats() const override;
+ void SetStereoChannelSwapping(bool enable) override;
+
VoiceEngine* voice_engine();
rtc::scoped_refptr<AudioMixer> mixer();
bool typing_noise_detected() const;
+ void AddSendingStream(webrtc::AudioSendStream* stream,
+ int sample_rate_hz, size_t num_channels);
+ void RemoveSendingStream(webrtc::AudioSendStream* stream);
+
private:
// rtc::RefCountInterface implementation.
void AddRef() const override;
rtc::RefCountReleaseStatus Release() const override;
+ void UpdateAudioTransportWithSendingStreams();
+
rtc::ThreadChecker thread_checker_;
rtc::ThreadChecker process_thread_checker_;
const webrtc::AudioState::Config config_;
@@ -63,14 +76,20 @@
mutable volatile int ref_count_ = 0;
// Transports mixed audio from the mixer to the audio device and
- // recorded audio to the VoE AudioTransport.
- AudioTransportProxy audio_transport_proxy_;
+ // recorded audio to the sending streams.
+ AudioTransportImpl audio_transport_;
// Null audio poller is used to continue polling the audio streams if audio
// playout is disabled so that audio processing still happens and the audio
// stats are still updated.
std::unique_ptr<NullAudioPoller> null_audio_poller_;
+ struct StreamProperties {
+ int sample_rate_hz = 0;
+ size_t num_channels = 0;
+ };
+ std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_;
+
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
};
} // namespace internal