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Add FixedGainController and move GainController2 in APM.

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The FixedGainController (FGC) applies a fixed gain. It will also
control the limiter. The limiter will be landed over the next several
CLs.

The GainController2 is a 'private submodule' of APM. It will control
the new automatic gain controller (AGC). It controls the AGC through
Initialize() and ApplyConfig().

This CL contains

* build changes to make modules/audio_processing/agc2 an independent
  target

* a new MutableFloatAudioFrame which is the audio interface between
  AGC2 and APM

* move of the fixed gain application from GainController2 to
  FixedGainController.

If you are a googler, there is more information in this doc:
https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit#

Bug: webrtc:7949
Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a
Reviewed-on: https://webrtc-review.googlesource.com/50440
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22046}
This commit is contained in:
Alex Loiko
2018-02-16 11:54:07 +01:00
committed by Commit Bot
parent 6df09f6f6a
commit e36e8bbf6d
20 changed files with 510 additions and 87 deletions
Download Patch File Download Diff File Expand all files Collapse all files

41
modules/audio_processing/include/aec_dump.h
View File

@ -16,6 +16,7 @@
#include <vector>
#include "api/array_view.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
@ -65,37 +66,6 @@ struct InternalAPMStreamsConfig {
size_t render_output_num_channels = 0;
};
// Class to pass audio data in float** format. This is to avoid
// dependence on AudioBuffer, and avoid problems associated with
// rtc::ArrayView<rtc::ArrayView>.
class FloatAudioFrame {
public:
// |num_channels| and |channel_size| describe the float**
// |audio_samples|. |audio_samples| is assumed to point to a
// two-dimensional |num_channels * channel_size| array of floats.
FloatAudioFrame(const float* const* audio_samples,
size_t num_channels,
size_t channel_size)
: audio_samples_(audio_samples),
num_channels_(num_channels),
channel_size_(channel_size) {}
FloatAudioFrame() = delete;
size_t num_channels() const { return num_channels_; }
rtc::ArrayView<const float> channel(size_t idx) const {
RTC_DCHECK_LE(0, idx);
RTC_DCHECK_LE(idx, num_channels_);
return rtc::ArrayView<const float>(audio_samples_[idx], channel_size_);
}
private:
const float* const* audio_samples_;
size_t num_channels_;
size_t channel_size_;
};
// An interface for recording configuration and input/output streams
// of the Audio Processing Module. The recordings are called
// 'aec-dumps' and are stored in a protobuf format defined in
@ -122,8 +92,10 @@ class AecDump {
// Logs Event::Type STREAM message. To log an input/output pair,
// call the AddCapture* and AddAudioProcessingState methods followed
// by a WriteCaptureStreamMessage call.
virtual void AddCaptureStreamInput(const FloatAudioFrame& src) = 0;
virtual void AddCaptureStreamOutput(const FloatAudioFrame& src) = 0;
virtual void AddCaptureStreamInput(
const AudioFrameView<const float>& src) = 0;
virtual void AddCaptureStreamOutput(
const AudioFrameView<const float>& src) = 0;
virtual void AddCaptureStreamInput(const AudioFrame& frame) = 0;
virtual void AddCaptureStreamOutput(const AudioFrame& frame) = 0;
virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
@ -131,7 +103,8 @@ class AecDump {
// Logs Event::Type REVERSE_STREAM message.
virtual void WriteRenderStreamMessage(const AudioFrame& frame) = 0;
virtual void WriteRenderStreamMessage(const FloatAudioFrame& src) = 0;
virtual void WriteRenderStreamMessage(
const AudioFrameView<const float>& src) = 0;
// Logs Event::Type CONFIG message.
virtual void WriteConfig(const InternalAPMConfig& config) = 0;