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/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index 6c19146..38ce742 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -104,6 +104,7 @@
"../../api:array_view",
"../../common_audio",
"../../rtc_base:checks",
+ "../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
"../../system_wrappers",
diff --git a/modules/audio_device/include/audio_device_defines.h b/modules/audio_device/include/audio_device_defines.h
index 510b07c..04119f6 100644
--- a/modules/audio_device/include/audio_device_defines.h
+++ b/modules/audio_device/include/audio_device_defines.h
@@ -13,6 +13,8 @@
#include <stddef.h>
+#include "rtc_base/checks.h"
+#include "rtc_base/deprecation.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
@@ -54,17 +56,19 @@
// The data will not undergo audio processing.
// |voe_channel| is the id of the VoE channel which is the sink to the
// capture data.
- virtual void PushCaptureData(int voe_channel,
- const void* audio_data,
- int bits_per_sample,
- int sample_rate,
- size_t number_of_channels,
- size_t number_of_frames) = 0;
+ // TODO(bugs.webrtc.org/8659): Remove this method once clients updated.
+ RTC_DEPRECATED virtual void PushCaptureData(
+ int voe_channel,
+ const void* audio_data,
+ int bits_per_sample,
+ int sample_rate,
+ size_t number_of_channels,
+ size_t number_of_frames) {
+ RTC_NOTREACHED();
+ }
// Method to pull mixed render audio data from all active VoE channels.
// The data will not be passed as reference for audio processing internally.
- // TODO(xians): Support getting the unmixed render data from specific VoE
- // channel.
virtual void PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,