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_transport_impl.h b/audio/audio_transport_impl.h
new file mode 100644
index 0000000..e7de7e9
--- /dev/null
+++ b/audio/audio_transport_impl.h
@@ -0,0 +1,100 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_TRANSPORT_IMPL_H_
+#define AUDIO_AUDIO_TRANSPORT_IMPL_H_
+
+#include <vector>
+
+#include "api/audio/audio_mixer.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/typing_detection.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/criticalsection.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "rtc_base/thread_annotations.h"
+#include "voice_engine/audio_level.h"
+
+namespace webrtc {
+
+class AudioSendStream;
+
+class AudioTransportImpl : public AudioTransport {
+ public:
+ AudioTransportImpl(AudioMixer* mixer,
+ AudioProcessing* audio_processing,
+ AudioDeviceModule* audio_device_module);
+ ~AudioTransportImpl() override;
+
+ int32_t RecordedDataIsAvailable(const void* audioSamples,
+ const size_t nSamples,
+ const size_t nBytesPerSample,
+ const size_t nChannels,
+ const uint32_t samplesPerSec,
+ const uint32_t totalDelayMS,
+ const int32_t clockDrift,
+ const uint32_t currentMicLevel,
+ const bool keyPressed,
+ uint32_t& newMicLevel) override;
+
+ int32_t NeedMorePlayData(const size_t nSamples,
+ const size_t nBytesPerSample,
+ const size_t nChannels,
+ const uint32_t samplesPerSec,
+ void* audioSamples,
+ size_t& nSamplesOut,
+ int64_t* elapsed_time_ms,
+ int64_t* ntp_time_ms) override;
+
+ void PullRenderData(int bits_per_sample,
+ int sample_rate,
+ size_t number_of_channels,
+ size_t number_of_frames,
+ void* audio_data,
+ int64_t* elapsed_time_ms,
+ int64_t* ntp_time_ms) override;
+
+ void UpdateSendingStreams(std::vector<AudioSendStream*> streams,
+ int send_sample_rate_hz, size_t send_num_channels);
+ void SetStereoChannelSwapping(bool enable);
+ bool typing_noise_detected() const;
+ const voe::AudioLevel& audio_level() const {
+ return audio_level_;
+ }
+
+ private:
+ // Shared.
+ AudioProcessing* audio_processing_ = nullptr;
+
+ // Capture side.
+ rtc::CriticalSection capture_lock_;
+ std::vector<AudioSendStream*> sending_streams_ RTC_GUARDED_BY(capture_lock_);
+ int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000;
+ size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
+ bool typing_noise_detected_ RTC_GUARDED_BY(capture_lock_) = false;
+ bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
+ AudioDeviceModule* audio_device_module_ = nullptr;
+ PushResampler<int16_t> capture_resampler_;
+ voe::AudioLevel audio_level_;
+ TypingDetection typing_detection_;
+
+ // Render side.
+ rtc::scoped_refptr<AudioMixer> mixer_;
+ AudioFrame mixed_frame_;
+ // Converts mixed audio to the audio device output rate.
+ PushResampler<int16_t> render_resampler_;
+
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportImpl);
+};
+} // namespace webrtc
+
+#endif // AUDIO_AUDIO_TRANSPORT_IMPL_H_
