mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
Move webrtc::AudioProcessing include to api/ folder
Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
This commit is contained in:
Florent Castelli
WebRTC LUCI CQ
parent
47393aabaa
commit
0afde7614d
116 changed files with 1219 additions and 1186 deletions
11
api/BUILD.gn
11
api/BUILD.gn
|
|
@ -56,8 +56,8 @@ rtc_source_set("enable_media_with_defaults") {
|
|||
deps = [
|
||||
":enable_media",
|
||||
":libjingle_peerconnection_api",
|
||||
"../modules/audio_processing:api",
|
||||
"../rtc_base/system:rtc_export",
|
||||
"audio:audio_processing",
|
||||
"audio_codecs:builtin_audio_decoder_factory",
|
||||
"audio_codecs:builtin_audio_encoder_factory",
|
||||
"task_queue:default_task_queue_factory",
|
||||
|
|
@ -81,13 +81,13 @@ if (!build_with_chromium) {
|
|||
":scoped_refptr",
|
||||
"../api/rtc_event_log:rtc_event_log_factory",
|
||||
"../modules/audio_device:audio_device_api",
|
||||
"../modules/audio_processing:api",
|
||||
"../pc:peer_connection_factory",
|
||||
"../pc:webrtc_sdp",
|
||||
"../rtc_base:threading",
|
||||
"../rtc_base/system:rtc_export",
|
||||
"../stats:rtc_stats",
|
||||
"audio:audio_mixer_api",
|
||||
"audio:audio_processing",
|
||||
"audio_codecs:audio_codecs_api",
|
||||
"task_queue:default_task_queue_factory",
|
||||
"transport:field_trial_based_config",
|
||||
|
|
@ -164,10 +164,10 @@ rtc_library("media_stream_interface") {
|
|||
":scoped_refptr",
|
||||
":sequence_checker",
|
||||
":video_track_source_constraints",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../rtc_base:checks",
|
||||
"../rtc_base/system:no_unique_address",
|
||||
"../rtc_base/system:rtc_export",
|
||||
"audio:audio_processing_statistics",
|
||||
"video:recordable_encoded_frame",
|
||||
"video:video_frame",
|
||||
]
|
||||
|
|
@ -377,7 +377,6 @@ rtc_library("libjingle_peerconnection_api") {
|
|||
# targets like pnacl. API should not depend on anything outside of this
|
||||
# file, really. All these should arguably go away in time.
|
||||
"../media:rtc_media_config",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../rtc_base:checks",
|
||||
"../rtc_base:ip_address",
|
||||
"../rtc_base:socket_address",
|
||||
|
|
@ -587,7 +586,6 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
|
|||
":track_id_stream_info_map",
|
||||
":video_quality_analyzer_api",
|
||||
"../media:media_constants",
|
||||
"../modules/audio_processing:api",
|
||||
"../rtc_base:checks",
|
||||
"../rtc_base:network",
|
||||
"../rtc_base:rtc_certificate_generator",
|
||||
|
|
@ -596,6 +594,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
|
|||
"../rtc_base:threading",
|
||||
"../test:fileutils",
|
||||
"audio:audio_mixer_api",
|
||||
"audio:audio_processing",
|
||||
"rtc_event_log",
|
||||
"task_queue",
|
||||
"test/pclf:media_configuration",
|
||||
|
|
@ -995,8 +994,8 @@ if (rtc_include_tests) {
|
|||
|
||||
deps = [
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/audio_processing:audioproc_f_impl",
|
||||
"audio:audio_processing",
|
||||
]
|
||||
}
|
||||
|
||||
|
|
|
|||
13
api/DEPS
13
api/DEPS
|
|
@ -49,6 +49,7 @@ specific_include_rules = {
|
|||
],
|
||||
|
||||
".*\.h": [
|
||||
"+rtc_base/arraysize.h",
|
||||
"+rtc_base/checks.h",
|
||||
"+rtc_base/system/rtc_export.h",
|
||||
"+rtc_base/system/rtc_export_template.h",
|
||||
|
|
@ -95,10 +96,6 @@ specific_include_rules = {
|
|||
"+modules/include/module_fec_types.h",
|
||||
],
|
||||
|
||||
"media_stream_interface\.h": [
|
||||
"+modules/audio_processing/include/audio_processing_statistics.h",
|
||||
],
|
||||
|
||||
"packet_socket_factory\.h": [
|
||||
"+rtc_base/proxy_info.h",
|
||||
"+rtc_base/async_packet_socket.h",
|
||||
|
|
@ -151,14 +148,6 @@ specific_include_rules = {
|
|||
"+rtc_base/buffer.h",
|
||||
],
|
||||
|
||||
"audioproc_float\.h": [
|
||||
"+modules/audio_processing/include/audio_processing.h",
|
||||
],
|
||||
|
||||
"echo_detector_creator\.h": [
|
||||
"+modules/audio_processing/include/audio_processing.h",
|
||||
],
|
||||
|
||||
"make_ref_counted\.h": [
|
||||
"+rtc_base/ref_counted_object.h",
|
||||
],
|
||||
|
|
|
|||
|
|
@ -42,6 +42,44 @@ rtc_source_set("audio_mixer_api") {
|
|||
]
|
||||
}
|
||||
|
||||
rtc_source_set("audio_processing") {
|
||||
visibility = [ "*" ]
|
||||
sources = [
|
||||
"audio_processing.cc",
|
||||
"audio_processing.h",
|
||||
]
|
||||
deps = [
|
||||
":aec3_config",
|
||||
":audio_processing_statistics",
|
||||
":echo_control",
|
||||
"..:array_view",
|
||||
"..:ref_count",
|
||||
"..:scoped_refptr",
|
||||
"../../rtc_base:checks",
|
||||
"../../rtc_base:macromagic",
|
||||
"../../rtc_base:stringutils",
|
||||
"../../rtc_base/system:arch",
|
||||
"../../rtc_base/system:file_wrapper",
|
||||
"../../rtc_base/system:rtc_export",
|
||||
"../task_queue",
|
||||
]
|
||||
absl_deps = [
|
||||
"//third_party/abseil-cpp/absl/base:nullability",
|
||||
"//third_party/abseil-cpp/absl/strings:string_view",
|
||||
"//third_party/abseil-cpp/absl/types:optional",
|
||||
]
|
||||
}
|
||||
|
||||
rtc_source_set("audio_processing_statistics") {
|
||||
visibility = [ "*" ]
|
||||
sources = [
|
||||
"audio_processing_statistics.cc",
|
||||
"audio_processing_statistics.h",
|
||||
]
|
||||
deps = [ "../../rtc_base/system:rtc_export" ]
|
||||
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
|
||||
}
|
||||
|
||||
rtc_library("aec3_config") {
|
||||
visibility = [ "*" ]
|
||||
sources = [
|
||||
|
|
@ -85,9 +123,9 @@ rtc_source_set("echo_detector_creator") {
|
|||
"echo_detector_creator.h",
|
||||
]
|
||||
deps = [
|
||||
":audio_processing",
|
||||
"..:make_ref_counted",
|
||||
"../../api:scoped_refptr",
|
||||
"../../modules/audio_processing:api",
|
||||
"../../modules/audio_processing:residual_echo_detector",
|
||||
]
|
||||
}
|
||||
|
|
|
|||
|
|
@ -8,10 +8,11 @@
|
|||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include <string>
|
||||
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/strings/string_builder.h"
|
||||
#include "rtc_base/system/arch.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace {
|
||||
945
api/audio/audio_processing.h
Normal file
945
api/audio/audio_processing.h
Normal file
|
|
@ -0,0 +1,945 @@
|
|||
/*
|
||||
* Copyright (c) 2012 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 API_AUDIO_AUDIO_PROCESSING_H_
|
||||
#define API_AUDIO_AUDIO_PROCESSING_H_
|
||||
|
||||
// MSVC++ requires this to be set before any other includes to get M_PI.
|
||||
#ifndef _USE_MATH_DEFINES
|
||||
#define _USE_MATH_DEFINES
|
||||
#endif
|
||||
|
||||
#include <math.h>
|
||||
#include <stddef.h> // size_t
|
||||
#include <stdio.h> // FILE
|
||||
#include <string.h>
|
||||
|
||||
#include <array>
|
||||
#include <cstdint>
|
||||
#include <memory>
|
||||
#include <string>
|
||||
#include <utility>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/audio/echo_control.h"
|
||||
#include "api/ref_count.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "api/task_queue/task_queue_base.h"
|
||||
#include "rtc_base/arraysize.h"
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/system/rtc_export.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
class AecDump;
|
||||
class AudioBuffer;
|
||||
|
||||
class StreamConfig;
|
||||
class ProcessingConfig;
|
||||
|
||||
class EchoDetector;
|
||||
class CustomAudioAnalyzer;
|
||||
class CustomProcessing;
|
||||
|
||||
// The Audio Processing Module (APM) provides a collection of voice processing
|
||||
// components designed for real-time communications software.
|
||||
//
|
||||
// APM operates on two audio streams on a frame-by-frame basis. Frames of the
|
||||
// primary stream, on which all processing is applied, are passed to
|
||||
// `ProcessStream()`. Frames of the reverse direction stream are passed to
|
||||
// `ProcessReverseStream()`. On the client-side, this will typically be the
|
||||
// near-end (capture) and far-end (render) streams, respectively. APM should be
|
||||
// placed in the signal chain as close to the audio hardware abstraction layer
|
||||
// (HAL) as possible.
|
||||
//
|
||||
// On the server-side, the reverse stream will normally not be used, with
|
||||
// processing occurring on each incoming stream.
|
||||
//
|
||||
// Component interfaces follow a similar pattern and are accessed through
|
||||
// corresponding getters in APM. All components are disabled at create-time,
|
||||
// with default settings that are recommended for most situations. New settings
|
||||
// can be applied without enabling a component. Enabling a component triggers
|
||||
// memory allocation and initialization to allow it to start processing the
|
||||
// streams.
|
||||
//
|
||||
// Thread safety is provided with the following assumptions to reduce locking
|
||||
// overhead:
|
||||
// 1. The stream getters and setters are called from the same thread as
|
||||
// ProcessStream(). More precisely, stream functions are never called
|
||||
// concurrently with ProcessStream().
|
||||
// 2. Parameter getters are never called concurrently with the corresponding
|
||||
// setter.
|
||||
//
|
||||
// APM accepts only linear PCM audio data in chunks of ~10 ms (see
|
||||
// AudioProcessing::GetFrameSize() for details) and sample rates ranging from
|
||||
// 8000 Hz to 384000 Hz. The int16 interfaces use interleaved data, while the
|
||||
// float interfaces use deinterleaved data.
|
||||
//
|
||||
// Usage example, omitting error checking:
|
||||
// rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
|
||||
//
|
||||
// AudioProcessing::Config config;
|
||||
// config.echo_canceller.enabled = true;
|
||||
// config.echo_canceller.mobile_mode = false;
|
||||
//
|
||||
// config.gain_controller1.enabled = true;
|
||||
// config.gain_controller1.mode =
|
||||
// AudioProcessing::Config::GainController1::kAdaptiveAnalog;
|
||||
// config.gain_controller1.analog_level_minimum = 0;
|
||||
// config.gain_controller1.analog_level_maximum = 255;
|
||||
//
|
||||
// config.gain_controller2.enabled = true;
|
||||
//
|
||||
// config.high_pass_filter.enabled = true;
|
||||
//
|
||||
// apm->ApplyConfig(config)
|
||||
//
|
||||
// // Start a voice call...
|
||||
//
|
||||
// // ... Render frame arrives bound for the audio HAL ...
|
||||
// apm->ProcessReverseStream(render_frame);
|
||||
//
|
||||
// // ... Capture frame arrives from the audio HAL ...
|
||||
// // Call required set_stream_ functions.
|
||||
// apm->set_stream_delay_ms(delay_ms);
|
||||
// apm->set_stream_analog_level(analog_level);
|
||||
//
|
||||
// apm->ProcessStream(capture_frame);
|
||||
//
|
||||
// // Call required stream_ functions.
|
||||
// analog_level = apm->recommended_stream_analog_level();
|
||||
// has_voice = apm->stream_has_voice();
|
||||
//
|
||||
// // Repeat render and capture processing for the duration of the call...
|
||||
// // Start a new call...
|
||||
// apm->Initialize();
|
||||
//
|
||||
// // Close the application...
|
||||
// apm.reset();
|
||||
//
|
||||
class RTC_EXPORT AudioProcessing : public RefCountInterface {
|
||||
public:
|
||||
// The struct below constitutes the new parameter scheme for the audio
|
||||
// processing. It is being introduced gradually and until it is fully
|
||||
// introduced, it is prone to change.
|
||||
// TODO(peah): Remove this comment once the new config scheme is fully rolled
|
||||
// out.
|
||||
//
|
||||
// The parameters and behavior of the audio processing module are controlled
|
||||
// by changing the default values in the AudioProcessing::Config struct.
|
||||
// The config is applied by passing the struct to the ApplyConfig method.
|
||||
//
|
||||
// This config is intended to be used during setup, and to enable/disable
|
||||
// top-level processing effects. Use during processing may cause undesired
|
||||
// submodule resets, affecting the audio quality. Use the RuntimeSetting
|
||||
// construct for runtime configuration.
|
||||
struct RTC_EXPORT Config {
|
||||
// Sets the properties of the audio processing pipeline.
|
||||
struct RTC_EXPORT Pipeline {
|
||||
// Ways to downmix a multi-channel track to mono.
|
||||
enum class DownmixMethod {
|
||||
kAverageChannels, // Average across channels.
|
||||
kUseFirstChannel // Use the first channel.
|
||||
};
|
||||
|
||||
// Maximum allowed processing rate used internally. May only be set to
|
||||
// 32000 or 48000 and any differing values will be treated as 48000.
|
||||
int maximum_internal_processing_rate = 48000;
|
||||
// Allow multi-channel processing of render audio.
|
||||
bool multi_channel_render = false;
|
||||
// Allow multi-channel processing of capture audio when AEC3 is active
|
||||
// or a custom AEC is injected..
|
||||
bool multi_channel_capture = false;
|
||||
// Indicates how to downmix multi-channel capture audio to mono (when
|
||||
// needed).
|
||||
DownmixMethod capture_downmix_method = DownmixMethod::kAverageChannels;
|
||||
} pipeline;
|
||||
|
||||
// Enabled the pre-amplifier. It amplifies the capture signal
|
||||
// before any other processing is done.
|
||||
// TODO(webrtc:5298): Deprecate and use the pre-gain functionality in
|
||||
// capture_level_adjustment instead.
|
||||
struct PreAmplifier {
|
||||
bool enabled = false;
|
||||
float fixed_gain_factor = 1.0f;
|
||||
} pre_amplifier;
|
||||
|
||||
// Functionality for general level adjustment in the capture pipeline. This
|
||||
// should not be used together with the legacy PreAmplifier functionality.
|
||||
struct CaptureLevelAdjustment {
|
||||
bool operator==(const CaptureLevelAdjustment& rhs) const;
|
||||
bool operator!=(const CaptureLevelAdjustment& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
bool enabled = false;
|
||||
// The `pre_gain_factor` scales the signal before any processing is done.
|
||||
float pre_gain_factor = 1.0f;
|
||||
// The `post_gain_factor` scales the signal after all processing is done.
|
||||
float post_gain_factor = 1.0f;
|
||||
struct AnalogMicGainEmulation {
|
||||
bool operator==(const AnalogMicGainEmulation& rhs) const;
|
||||
bool operator!=(const AnalogMicGainEmulation& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
bool enabled = false;
|
||||
// Initial analog gain level to use for the emulated analog gain. Must
|
||||
// be in the range [0...255].
|
||||
int initial_level = 255;
|
||||
} analog_mic_gain_emulation;
|
||||
} capture_level_adjustment;
|
||||
|
||||
struct HighPassFilter {
|
||||
bool enabled = false;
|
||||
bool apply_in_full_band = true;
|
||||
} high_pass_filter;
|
||||
|
||||
struct EchoCanceller {
|
||||
bool enabled = false;
|
||||
bool mobile_mode = false;
|
||||
bool export_linear_aec_output = false;
|
||||
// Enforce the highpass filter to be on (has no effect for the mobile
|
||||
// mode).
|
||||
bool enforce_high_pass_filtering = true;
|
||||
} echo_canceller;
|
||||
|
||||
// Enables background noise suppression.
|
||||
struct NoiseSuppression {
|
||||
bool enabled = false;
|
||||
enum Level { kLow, kModerate, kHigh, kVeryHigh };
|
||||
Level level = kModerate;
|
||||
bool analyze_linear_aec_output_when_available = false;
|
||||
} noise_suppression;
|
||||
|
||||
// Enables transient suppression.
|
||||
struct TransientSuppression {
|
||||
bool enabled = false;
|
||||
} transient_suppression;
|
||||
|
||||
// Enables automatic gain control (AGC) functionality.
|
||||
// The automatic gain control (AGC) component brings the signal to an
|
||||
// appropriate range. This is done by applying a digital gain directly and,
|
||||
// in the analog mode, prescribing an analog gain to be applied at the audio
|
||||
// HAL.
|
||||
// Recommended to be enabled on the client-side.
|
||||
struct RTC_EXPORT GainController1 {
|
||||
bool operator==(const GainController1& rhs) const;
|
||||
bool operator!=(const GainController1& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
|
||||
bool enabled = false;
|
||||
enum Mode {
|
||||
// Adaptive mode intended for use if an analog volume control is
|
||||
// available on the capture device. It will require the user to provide
|
||||
// coupling between the OS mixer controls and AGC through the
|
||||
// stream_analog_level() functions.
|
||||
// It consists of an analog gain prescription for the audio device and a
|
||||
// digital compression stage.
|
||||
kAdaptiveAnalog,
|
||||
// Adaptive mode intended for situations in which an analog volume
|
||||
// control is unavailable. It operates in a similar fashion to the
|
||||
// adaptive analog mode, but with scaling instead applied in the digital
|
||||
// domain. As with the analog mode, it additionally uses a digital
|
||||
// compression stage.
|
||||
kAdaptiveDigital,
|
||||
// Fixed mode which enables only the digital compression stage also used
|
||||
// by the two adaptive modes.
|
||||
// It is distinguished from the adaptive modes by considering only a
|
||||
// short time-window of the input signal. It applies a fixed gain
|
||||
// through most of the input level range, and compresses (gradually
|
||||
// reduces gain with increasing level) the input signal at higher
|
||||
// levels. This mode is preferred on embedded devices where the capture
|
||||
// signal level is predictable, so that a known gain can be applied.
|
||||
kFixedDigital
|
||||
};
|
||||
Mode mode = kAdaptiveAnalog;
|
||||
// Sets the target peak level (or envelope) of the AGC in dBFs (decibels
|
||||
// from digital full-scale). The convention is to use positive values. For
|
||||
// instance, passing in a value of 3 corresponds to -3 dBFs, or a target
|
||||
// level 3 dB below full-scale. Limited to [0, 31].
|
||||
int target_level_dbfs = 3;
|
||||
// Sets the maximum gain the digital compression stage may apply, in dB. A
|
||||
// higher number corresponds to greater compression, while a value of 0
|
||||
// will leave the signal uncompressed. Limited to [0, 90].
|
||||
// For updates after APM setup, use a RuntimeSetting instead.
|
||||
int compression_gain_db = 9;
|
||||
// When enabled, the compression stage will hard limit the signal to the
|
||||
// target level. Otherwise, the signal will be compressed but not limited
|
||||
// above the target level.
|
||||
bool enable_limiter = true;
|
||||
|
||||
// Enables the analog gain controller functionality.
|
||||
struct AnalogGainController {
|
||||
bool enabled = true;
|
||||
// TODO(bugs.webrtc.org/7494): Deprecated. Stop using and remove.
|
||||
int startup_min_volume = 0;
|
||||
// Lowest analog microphone level that will be applied in response to
|
||||
// clipping.
|
||||
int clipped_level_min = 70;
|
||||
// If true, an adaptive digital gain is applied.
|
||||
bool enable_digital_adaptive = true;
|
||||
// Amount the microphone level is lowered with every clipping event.
|
||||
// Limited to (0, 255].
|
||||
int clipped_level_step = 15;
|
||||
// Proportion of clipped samples required to declare a clipping event.
|
||||
// Limited to (0.f, 1.f).
|
||||
float clipped_ratio_threshold = 0.1f;
|
||||
// Time in frames to wait after a clipping event before checking again.
|
||||
// Limited to values higher than 0.
|
||||
int clipped_wait_frames = 300;
|
||||
|
||||
// Enables clipping prediction functionality.
|
||||
struct ClippingPredictor {
|
||||
bool enabled = false;
|
||||
enum Mode {
|
||||
// Clipping event prediction mode with fixed step estimation.
|
||||
kClippingEventPrediction,
|
||||
// Clipped peak estimation mode with adaptive step estimation.
|
||||
kAdaptiveStepClippingPeakPrediction,
|
||||
// Clipped peak estimation mode with fixed step estimation.
|
||||
kFixedStepClippingPeakPrediction,
|
||||
};
|
||||
Mode mode = kClippingEventPrediction;
|
||||
// Number of frames in the sliding analysis window.
|
||||
int window_length = 5;
|
||||
// Number of frames in the sliding reference window.
|
||||
int reference_window_length = 5;
|
||||
// Reference window delay (unit: number of frames).
|
||||
int reference_window_delay = 5;
|
||||
// Clipping prediction threshold (dBFS).
|
||||
float clipping_threshold = -1.0f;
|
||||
// Crest factor drop threshold (dB).
|
||||
float crest_factor_margin = 3.0f;
|
||||
// If true, the recommended clipped level step is used to modify the
|
||||
// analog gain. Otherwise, the predictor runs without affecting the
|
||||
// analog gain.
|
||||
bool use_predicted_step = true;
|
||||
} clipping_predictor;
|
||||
} analog_gain_controller;
|
||||
} gain_controller1;
|
||||
|
||||
// Parameters for AGC2, an Automatic Gain Control (AGC) sub-module which
|
||||
// replaces the AGC sub-module parametrized by `gain_controller1`.
|
||||
// AGC2 brings the captured audio signal to the desired level by combining
|
||||
// three different controllers (namely, input volume controller, adapative
|
||||
// digital controller and fixed digital controller) and a limiter.
|
||||
// TODO(bugs.webrtc.org:7494): Name `GainController` when AGC1 removed.
|
||||
struct RTC_EXPORT GainController2 {
|
||||
bool operator==(const GainController2& rhs) const;
|
||||
bool operator!=(const GainController2& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
|
||||
// AGC2 must be created if and only if `enabled` is true.
|
||||
bool enabled = false;
|
||||
|
||||
// Parameters for the input volume controller, which adjusts the input
|
||||
// volume applied when the audio is captured (e.g., microphone volume on
|
||||
// a soundcard, input volume on HAL).
|
||||
struct InputVolumeController {
|
||||
bool operator==(const InputVolumeController& rhs) const;
|
||||
bool operator!=(const InputVolumeController& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
bool enabled = false;
|
||||
} input_volume_controller;
|
||||
|
||||
// Parameters for the adaptive digital controller, which adjusts and
|
||||
// applies a digital gain after echo cancellation and after noise
|
||||
// suppression.
|
||||
struct RTC_EXPORT AdaptiveDigital {
|
||||
bool operator==(const AdaptiveDigital& rhs) const;
|
||||
bool operator!=(const AdaptiveDigital& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
bool enabled = false;
|
||||
float headroom_db = 5.0f;
|
||||
float max_gain_db = 50.0f;
|
||||
float initial_gain_db = 15.0f;
|
||||
float max_gain_change_db_per_second = 6.0f;
|
||||
float max_output_noise_level_dbfs = -50.0f;
|
||||
} adaptive_digital;
|
||||
|
||||
// Parameters for the fixed digital controller, which applies a fixed
|
||||
// digital gain after the adaptive digital controller and before the
|
||||
// limiter.
|
||||
struct FixedDigital {
|
||||
// By setting `gain_db` to a value greater than zero, the limiter can be
|
||||
// turned into a compressor that first applies a fixed gain.
|
||||
float gain_db = 0.0f;
|
||||
} fixed_digital;
|
||||
} gain_controller2;
|
||||
|
||||
std::string ToString() const;
|
||||
};
|
||||
|
||||
// Specifies the properties of a setting to be passed to AudioProcessing at
|
||||
// runtime.
|
||||
class RuntimeSetting {
|
||||
public:
|
||||
enum class Type {
|
||||
kNotSpecified,
|
||||
kCapturePreGain,
|
||||
kCaptureCompressionGain,
|
||||
kCaptureFixedPostGain,
|
||||
kPlayoutVolumeChange,
|
||||
kCustomRenderProcessingRuntimeSetting,
|
||||
kPlayoutAudioDeviceChange,
|
||||
kCapturePostGain,
|
||||
kCaptureOutputUsed
|
||||
};
|
||||
|
||||
// Play-out audio device properties.
|
||||
struct PlayoutAudioDeviceInfo {
|
||||
int id; // Identifies the audio device.
|
||||
int max_volume; // Maximum play-out volume.
|
||||
};
|
||||
|
||||
RuntimeSetting() : type_(Type::kNotSpecified), value_(0.0f) {}
|
||||
~RuntimeSetting() = default;
|
||||
|
||||
static RuntimeSetting CreateCapturePreGain(float gain) {
|
||||
return {Type::kCapturePreGain, gain};
|
||||
}
|
||||
|
||||
static RuntimeSetting CreateCapturePostGain(float gain) {
|
||||
return {Type::kCapturePostGain, gain};
|
||||
}
|
||||
|
||||
// Corresponds to Config::GainController1::compression_gain_db, but for
|
||||
// runtime configuration.
|
||||
static RuntimeSetting CreateCompressionGainDb(int gain_db) {
|
||||
RTC_DCHECK_GE(gain_db, 0);
|
||||
RTC_DCHECK_LE(gain_db, 90);
|
||||
return {Type::kCaptureCompressionGain, static_cast<float>(gain_db)};
|
||||
}
|
||||
|
||||
// Corresponds to Config::GainController2::fixed_digital::gain_db, but for
|
||||
// runtime configuration.
|
||||
static RuntimeSetting CreateCaptureFixedPostGain(float gain_db) {
|
||||
RTC_DCHECK_GE(gain_db, 0.0f);
|
||||
RTC_DCHECK_LE(gain_db, 90.0f);
|
||||
return {Type::kCaptureFixedPostGain, gain_db};
|
||||
}
|
||||
|
||||
// Creates a runtime setting to notify play-out (aka render) audio device
|
||||
// changes.
|
||||
static RuntimeSetting CreatePlayoutAudioDeviceChange(
|
||||
PlayoutAudioDeviceInfo audio_device) {
|
||||
return {Type::kPlayoutAudioDeviceChange, audio_device};
|
||||
}
|
||||
|
||||
// Creates a runtime setting to notify play-out (aka render) volume changes.
|
||||
// `volume` is the unnormalized volume, the maximum of which
|
||||
static RuntimeSetting CreatePlayoutVolumeChange(int volume) {
|
||||
return {Type::kPlayoutVolumeChange, volume};
|
||||
}
|
||||
|
||||
static RuntimeSetting CreateCustomRenderSetting(float payload) {
|
||||
return {Type::kCustomRenderProcessingRuntimeSetting, payload};
|
||||
}
|
||||
|
||||
static RuntimeSetting CreateCaptureOutputUsedSetting(
|
||||
bool capture_output_used) {
|
||||
return {Type::kCaptureOutputUsed, capture_output_used};
|
||||
}
|
||||
|
||||
Type type() const { return type_; }
|
||||
// Getters do not return a value but instead modify the argument to protect
|
||||
// from implicit casting.
|
||||
void GetFloat(float* value) const {
|
||||
RTC_DCHECK(value);
|
||||
*value = value_.float_value;
|
||||
}
|
||||
void GetInt(int* value) const {
|
||||
RTC_DCHECK(value);
|
||||
*value = value_.int_value;
|
||||
}
|
||||
void GetBool(bool* value) const {
|
||||
RTC_DCHECK(value);
|
||||
*value = value_.bool_value;
|
||||
}
|
||||
void GetPlayoutAudioDeviceInfo(PlayoutAudioDeviceInfo* value) const {
|
||||
RTC_DCHECK(value);
|
||||
*value = value_.playout_audio_device_info;
|
||||
}
|
||||
|
||||
private:
|
||||
RuntimeSetting(Type id, float value) : type_(id), value_(value) {}
|
||||
RuntimeSetting(Type id, int value) : type_(id), value_(value) {}
|
||||
RuntimeSetting(Type id, PlayoutAudioDeviceInfo value)
|
||||
: type_(id), value_(value) {}
|
||||
Type type_;
|
||||
union U {
|
||||
U() {}
|
||||
U(int value) : int_value(value) {}
|
||||
U(float value) : float_value(value) {}
|
||||
U(PlayoutAudioDeviceInfo value) : playout_audio_device_info(value) {}
|
||||
float float_value;
|
||||
int int_value;
|
||||
bool bool_value;
|
||||
PlayoutAudioDeviceInfo playout_audio_device_info;
|
||||
} value_;
|
||||
};
|
||||
|
||||
~AudioProcessing() override {}
|
||||
|
||||
// Initializes internal states, while retaining all user settings. This
|
||||
// should be called before beginning to process a new audio stream. However,
|
||||
// it is not necessary to call before processing the first stream after
|
||||
// creation.
|
||||
//
|
||||
// It is also not necessary to call if the audio parameters (sample
|
||||
// rate and number of channels) have changed. Passing updated parameters
|
||||
// directly to `ProcessStream()` and `ProcessReverseStream()` is permissible.
|
||||
// If the parameters are known at init-time though, they may be provided.
|
||||
// TODO(webrtc:5298): Change to return void.
|
||||
virtual int Initialize() = 0;
|
||||
|
||||
// The int16 interfaces require:
|
||||
// - only `NativeRate`s be used
|
||||
// - that the input, output and reverse rates must match
|
||||
// - that `processing_config.output_stream()` matches
|
||||
// `processing_config.input_stream()`.
|
||||
//
|
||||
// The float interfaces accept arbitrary rates and support differing input and
|
||||
// output layouts, but the output must have either one channel or the same
|
||||
// number of channels as the input.
|
||||
virtual int Initialize(const ProcessingConfig& processing_config) = 0;
|
||||
|
||||
// TODO(peah): This method is a temporary solution used to take control
|
||||
// over the parameters in the audio processing module and is likely to change.
|
||||
virtual void ApplyConfig(const Config& config) = 0;
|
||||
|
||||
// TODO(ajm): Only intended for internal use. Make private and friend the
|
||||
// necessary classes?
|
||||
virtual int proc_sample_rate_hz() const = 0;
|
||||
virtual int proc_split_sample_rate_hz() const = 0;
|
||||
virtual size_t num_input_channels() const = 0;
|
||||
virtual size_t num_proc_channels() const = 0;
|
||||
virtual size_t num_output_channels() const = 0;
|
||||
virtual size_t num_reverse_channels() const = 0;
|
||||
|
||||
// Set to true when the output of AudioProcessing will be muted or in some
|
||||
// other way not used. Ideally, the captured audio would still be processed,
|
||||
// but some components may change behavior based on this information.
|
||||
// Default false. This method takes a lock. To achieve this in a lock-less
|
||||
// manner the PostRuntimeSetting can instead be used.
|
||||
virtual void set_output_will_be_muted(bool muted) = 0;
|
||||
|
||||
// Enqueues a runtime setting.
|
||||
virtual void SetRuntimeSetting(RuntimeSetting setting) = 0;
|
||||
|
||||
// Enqueues a runtime setting. Returns a bool indicating whether the
|
||||
// enqueueing was successfull.
|
||||
virtual bool PostRuntimeSetting(RuntimeSetting setting) = 0;
|
||||
|
||||
// Accepts and produces a ~10 ms frame of interleaved 16 bit integer audio as
|
||||
// specified in `input_config` and `output_config`. `src` and `dest` may use
|
||||
// the same memory, if desired.
|
||||
virtual int ProcessStream(const int16_t* const src,
|
||||
const StreamConfig& input_config,
|
||||
const StreamConfig& output_config,
|
||||
int16_t* const dest) = 0;
|
||||
|
||||
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
|
||||
// `src` points to a channel buffer, arranged according to `input_stream`. At
|
||||
// output, the channels will be arranged according to `output_stream` in
|
||||
// `dest`.
|
||||
//
|
||||
// The output must have one channel or as many channels as the input. `src`
|
||||
// and `dest` may use the same memory, if desired.
|
||||
virtual int ProcessStream(const float* const* src,
|
||||
const StreamConfig& input_config,
|
||||
const StreamConfig& output_config,
|
||||
float* const* dest) = 0;
|
||||
|
||||
// Accepts and produces a ~10 ms frame of interleaved 16 bit integer audio for
|
||||
// the reverse direction audio stream as specified in `input_config` and
|
||||
// `output_config`. `src` and `dest` may use the same memory, if desired.
|
||||
virtual int ProcessReverseStream(const int16_t* const src,
|
||||
const StreamConfig& input_config,
|
||||
const StreamConfig& output_config,
|
||||
int16_t* const dest) = 0;
|
||||
|
||||
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
|
||||
// `data` points to a channel buffer, arranged according to `reverse_config`.
|
||||
virtual int ProcessReverseStream(const float* const* src,
|
||||
const StreamConfig& input_config,
|
||||
const StreamConfig& output_config,
|
||||
float* const* dest) = 0;
|
||||
|
||||
// Accepts deinterleaved float audio with the range [-1, 1]. Each element
|
||||
// of `data` points to a channel buffer, arranged according to
|
||||
// `reverse_config`.
|
||||
virtual int AnalyzeReverseStream(const float* const* data,
|
||||
const StreamConfig& reverse_config) = 0;
|
||||
|
||||
// Returns the most recently produced ~10 ms of the linear AEC output at a
|
||||
// rate of 16 kHz. If there is more than one capture channel, a mono
|
||||
// representation of the input is returned. Returns true/false to indicate
|
||||
// whether an output returned.
|
||||
virtual bool GetLinearAecOutput(
|
||||
rtc::ArrayView<std::array<float, 160>> linear_output) const = 0;
|
||||
|
||||
// This must be called prior to ProcessStream() if and only if adaptive analog
|
||||
// gain control is enabled, to pass the current analog level from the audio
|
||||
// HAL. Must be within the range [0, 255].
|
||||
virtual void set_stream_analog_level(int level) = 0;
|
||||
|
||||
// When an analog mode is set, this should be called after
|
||||
// `set_stream_analog_level()` and `ProcessStream()` to obtain the recommended
|
||||
// new analog level for the audio HAL. It is the user's responsibility to
|
||||
// apply this level.
|
||||
virtual int recommended_stream_analog_level() const = 0;
|
||||
|
||||
// This must be called if and only if echo processing is enabled.
|
||||
//
|
||||
// Sets the `delay` in ms between ProcessReverseStream() receiving a far-end
|
||||
// frame and ProcessStream() receiving a near-end frame containing the
|
||||
// corresponding echo. On the client-side this can be expressed as
|
||||
// delay = (t_render - t_analyze) + (t_process - t_capture)
|
||||
// where,
|
||||
// - t_analyze is the time a frame is passed to ProcessReverseStream() and
|
||||
// t_render is the time the first sample of the same frame is rendered by
|
||||
// the audio hardware.
|
||||
// - t_capture is the time the first sample of a frame is captured by the
|
||||
// audio hardware and t_process is the time the same frame is passed to
|
||||
// ProcessStream().
|
||||
virtual int set_stream_delay_ms(int delay) = 0;
|
||||
virtual int stream_delay_ms() const = 0;
|
||||
|
||||
// Call to signal that a key press occurred (true) or did not occur (false)
|
||||
// with this chunk of audio.
|
||||
virtual void set_stream_key_pressed(bool key_pressed) = 0;
|
||||
|
||||
// Creates and attaches an webrtc::AecDump for recording debugging
|
||||
// information.
|
||||
// The `worker_queue` may not be null and must outlive the created
|
||||
// AecDump instance. |max_log_size_bytes == -1| means the log size
|
||||
// will be unlimited. `handle` may not be null. The AecDump takes
|
||||
// responsibility for `handle` and closes it in the destructor. A
|
||||
// return value of true indicates that the file has been
|
||||
// sucessfully opened, while a value of false indicates that
|
||||
// opening the file failed.
|
||||
virtual bool CreateAndAttachAecDump(
|
||||
absl::string_view file_name,
|
||||
int64_t max_log_size_bytes,
|
||||
absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
|
||||
virtual bool CreateAndAttachAecDump(
|
||||
absl::Nonnull<FILE*> handle,
|
||||
int64_t max_log_size_bytes,
|
||||
absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
|
||||
|
||||
// TODO(webrtc:5298) Deprecated variant.
|
||||
// Attaches provided webrtc::AecDump for recording debugging
|
||||
// information. Log file and maximum file size logic is supposed to
|
||||
// be handled by implementing instance of AecDump. Calling this
|
||||
// method when another AecDump is attached resets the active AecDump
|
||||
// with a new one. This causes the d-tor of the earlier AecDump to
|
||||
// be called. The d-tor call may block until all pending logging
|
||||
// tasks are completed.
|
||||
virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) = 0;
|
||||
|
||||
// If no AecDump is attached, this has no effect. If an AecDump is
|
||||
// attached, it's destructor is called. The d-tor may block until
|
||||
// all pending logging tasks are completed.
|
||||
virtual void DetachAecDump() = 0;
|
||||
|
||||
// Get audio processing statistics.
|
||||
virtual AudioProcessingStats GetStatistics() = 0;
|
||||
// TODO(webrtc:5298) Deprecated variant. The `has_remote_tracks` argument
|
||||
// should be set if there are active remote tracks (this would usually be true
|
||||
// during a call). If there are no remote tracks some of the stats will not be
|
||||
// set by AudioProcessing, because they only make sense if there is at least
|
||||
// one remote track.
|
||||
virtual AudioProcessingStats GetStatistics(bool has_remote_tracks) = 0;
|
||||
|
||||
// Returns the last applied configuration.
|
||||
virtual AudioProcessing::Config GetConfig() const = 0;
|
||||
|
||||
enum Error {
|
||||
// Fatal errors.
|
||||
kNoError = 0,
|
||||
kUnspecifiedError = -1,
|
||||
kCreationFailedError = -2,
|
||||
kUnsupportedComponentError = -3,
|
||||
kUnsupportedFunctionError = -4,
|
||||
kNullPointerError = -5,
|
||||
kBadParameterError = -6,
|
||||
kBadSampleRateError = -7,
|
||||
kBadDataLengthError = -8,
|
||||
kBadNumberChannelsError = -9,
|
||||
kFileError = -10,
|
||||
kStreamParameterNotSetError = -11,
|
||||
kNotEnabledError = -12,
|
||||
|
||||
// Warnings are non-fatal.
|
||||
// This results when a set_stream_ parameter is out of range. Processing
|
||||
// will continue, but the parameter may have been truncated.
|
||||
kBadStreamParameterWarning = -13
|
||||
};
|
||||
|
||||
// Native rates supported by the integer interfaces.
|
||||
enum NativeRate {
|
||||
kSampleRate8kHz = 8000,
|
||||
kSampleRate16kHz = 16000,
|
||||
kSampleRate32kHz = 32000,
|
||||
kSampleRate48kHz = 48000
|
||||
};
|
||||
|
||||
// TODO(kwiberg): We currently need to support a compiler (Visual C++) that
|
||||
// complains if we don't explicitly state the size of the array here. Remove
|
||||
// the size when that's no longer the case.
|
||||
static constexpr int kNativeSampleRatesHz[4] = {
|
||||
kSampleRate8kHz, kSampleRate16kHz, kSampleRate32kHz, kSampleRate48kHz};
|
||||
static constexpr size_t kNumNativeSampleRates =
|
||||
arraysize(kNativeSampleRatesHz);
|
||||
static constexpr int kMaxNativeSampleRateHz =
|
||||
kNativeSampleRatesHz[kNumNativeSampleRates - 1];
|
||||
|
||||
// APM processes audio in chunks of about 10 ms. See GetFrameSize() for
|
||||
// details.
|
||||
static constexpr int kChunkSizeMs = 10;
|
||||
|
||||
// Returns floor(sample_rate_hz/100): the number of samples per channel used
|
||||
// as input and output to the audio processing module in calls to
|
||||
// ProcessStream, ProcessReverseStream, AnalyzeReverseStream, and
|
||||
// GetLinearAecOutput.
|
||||
//
|
||||
// This is exactly 10 ms for sample rates divisible by 100. For example:
|
||||
// - 48000 Hz (480 samples per channel),
|
||||
// - 44100 Hz (441 samples per channel),
|
||||
// - 16000 Hz (160 samples per channel).
|
||||
//
|
||||
// Sample rates not divisible by 100 are received/produced in frames of
|
||||
// approximately 10 ms. For example:
|
||||
// - 22050 Hz (220 samples per channel, or ~9.98 ms per frame),
|
||||
// - 11025 Hz (110 samples per channel, or ~9.98 ms per frame).
|
||||
// These nondivisible sample rates yield lower audio quality compared to
|
||||
// multiples of 100. Internal resampling to 10 ms frames causes a simulated
|
||||
// clock drift effect which impacts the performance of (for example) echo
|
||||
// cancellation.
|
||||
static int GetFrameSize(int sample_rate_hz) { return sample_rate_hz / 100; }
|
||||
};
|
||||
|
||||
class RTC_EXPORT AudioProcessingBuilder {
|
||||
public:
|
||||
AudioProcessingBuilder();
|
||||
AudioProcessingBuilder(const AudioProcessingBuilder&) = delete;
|
||||
AudioProcessingBuilder& operator=(const AudioProcessingBuilder&) = delete;
|
||||
~AudioProcessingBuilder();
|
||||
|
||||
// Sets the APM configuration.
|
||||
AudioProcessingBuilder& SetConfig(const AudioProcessing::Config& config) {
|
||||
config_ = config;
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the echo controller factory to inject when APM is created.
|
||||
AudioProcessingBuilder& SetEchoControlFactory(
|
||||
std::unique_ptr<EchoControlFactory> echo_control_factory) {
|
||||
echo_control_factory_ = std::move(echo_control_factory);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the capture post-processing sub-module to inject when APM is created.
|
||||
AudioProcessingBuilder& SetCapturePostProcessing(
|
||||
std::unique_ptr<CustomProcessing> capture_post_processing) {
|
||||
capture_post_processing_ = std::move(capture_post_processing);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the render pre-processing sub-module to inject when APM is created.
|
||||
AudioProcessingBuilder& SetRenderPreProcessing(
|
||||
std::unique_ptr<CustomProcessing> render_pre_processing) {
|
||||
render_pre_processing_ = std::move(render_pre_processing);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the echo detector to inject when APM is created.
|
||||
AudioProcessingBuilder& SetEchoDetector(
|
||||
rtc::scoped_refptr<EchoDetector> echo_detector) {
|
||||
echo_detector_ = std::move(echo_detector);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the capture analyzer sub-module to inject when APM is created.
|
||||
AudioProcessingBuilder& SetCaptureAnalyzer(
|
||||
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer) {
|
||||
capture_analyzer_ = std::move(capture_analyzer);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Creates an APM instance with the specified config or the default one if
|
||||
// unspecified. Injects the specified components transferring the ownership
|
||||
// to the newly created APM instance - i.e., except for the config, the
|
||||
// builder is reset to its initial state.
|
||||
rtc::scoped_refptr<AudioProcessing> Create();
|
||||
|
||||
private:
|
||||
AudioProcessing::Config config_;
|
||||
std::unique_ptr<EchoControlFactory> echo_control_factory_;
|
||||
std::unique_ptr<CustomProcessing> capture_post_processing_;
|
||||
std::unique_ptr<CustomProcessing> render_pre_processing_;
|
||||
rtc::scoped_refptr<EchoDetector> echo_detector_;
|
||||
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer_;
|
||||
};
|
||||
|
||||
class StreamConfig {
|
||||
public:
|
||||
// sample_rate_hz: The sampling rate of the stream.
|
||||
// num_channels: The number of audio channels in the stream.
|
||||
StreamConfig(int sample_rate_hz = 0,
|
||||
size_t num_channels = 0) // NOLINT(runtime/explicit)
|
||||
: sample_rate_hz_(sample_rate_hz),
|
||||
num_channels_(num_channels),
|
||||
num_frames_(calculate_frames(sample_rate_hz)) {}
|
||||
|
||||
void set_sample_rate_hz(int value) {
|
||||
sample_rate_hz_ = value;
|
||||
num_frames_ = calculate_frames(value);
|
||||
}
|
||||
void set_num_channels(size_t value) { num_channels_ = value; }
|
||||
|
||||
int sample_rate_hz() const { return sample_rate_hz_; }
|
||||
|
||||
// The number of channels in the stream.
|
||||
size_t num_channels() const { return num_channels_; }
|
||||
|
||||
size_t num_frames() const { return num_frames_; }
|
||||
size_t num_samples() const { return num_channels_ * num_frames_; }
|
||||
|
||||
bool operator==(const StreamConfig& other) const {
|
||||
return sample_rate_hz_ == other.sample_rate_hz_ &&
|
||||
num_channels_ == other.num_channels_;
|
||||
}
|
||||
|
||||
bool operator!=(const StreamConfig& other) const { return !(*this == other); }
|
||||
|
||||
private:
|
||||
static size_t calculate_frames(int sample_rate_hz) {
|
||||
return static_cast<size_t>(AudioProcessing::GetFrameSize(sample_rate_hz));
|
||||
}
|
||||
|
||||
int sample_rate_hz_;
|
||||
size_t num_channels_;
|
||||
size_t num_frames_;
|
||||
};
|
||||
|
||||
class ProcessingConfig {
|
||||
public:
|
||||
enum StreamName {
|
||||
kInputStream,
|
||||
kOutputStream,
|
||||
kReverseInputStream,
|
||||
kReverseOutputStream,
|
||||
kNumStreamNames,
|
||||
};
|
||||
|
||||
const StreamConfig& input_stream() const {
|
||||
return streams[StreamName::kInputStream];
|
||||
}
|
||||
const StreamConfig& output_stream() const {
|
||||
return streams[StreamName::kOutputStream];
|
||||
}
|
||||
const StreamConfig& reverse_input_stream() const {
|
||||
return streams[StreamName::kReverseInputStream];
|
||||
}
|
||||
const StreamConfig& reverse_output_stream() const {
|
||||
return streams[StreamName::kReverseOutputStream];
|
||||
}
|
||||
|
||||
StreamConfig& input_stream() { return streams[StreamName::kInputStream]; }
|
||||
StreamConfig& output_stream() { return streams[StreamName::kOutputStream]; }
|
||||
StreamConfig& reverse_input_stream() {
|
||||
return streams[StreamName::kReverseInputStream];
|
||||
}
|
||||
StreamConfig& reverse_output_stream() {
|
||||
return streams[StreamName::kReverseOutputStream];
|
||||
}
|
||||
|
||||
bool operator==(const ProcessingConfig& other) const {
|
||||
for (int i = 0; i < StreamName::kNumStreamNames; ++i) {
|
||||
if (this->streams[i] != other.streams[i]) {
|
||||
return false;
|
||||
}
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
||||
bool operator!=(const ProcessingConfig& other) const {
|
||||
return !(*this == other);
|
||||
}
|
||||
|
||||
StreamConfig streams[StreamName::kNumStreamNames];
|
||||
};
|
||||
|
||||
// Experimental interface for a custom analysis submodule.
|
||||
class CustomAudioAnalyzer {
|
||||
public:
|
||||
// (Re-) Initializes the submodule.
|
||||
virtual void Initialize(int sample_rate_hz, int num_channels) = 0;
|
||||
// Analyzes the given capture or render signal.
|
||||
virtual void Analyze(const AudioBuffer* audio) = 0;
|
||||
// Returns a string representation of the module state.
|
||||
virtual std::string ToString() const = 0;
|
||||
|
||||
virtual ~CustomAudioAnalyzer() {}
|
||||
};
|
||||
|
||||
// Interface for a custom processing submodule.
|
||||
class CustomProcessing {
|
||||
public:
|
||||
// (Re-)Initializes the submodule.
|
||||
virtual void Initialize(int sample_rate_hz, int num_channels) = 0;
|
||||
// Processes the given capture or render signal.
|
||||
virtual void Process(AudioBuffer* audio) = 0;
|
||||
// Returns a string representation of the module state.
|
||||
virtual std::string ToString() const = 0;
|
||||
// Handles RuntimeSettings. TODO(webrtc:9262): make pure virtual
|
||||
// after updating dependencies.
|
||||
virtual void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting);
|
||||
|
||||
virtual ~CustomProcessing() {}
|
||||
};
|
||||
|
||||
// Interface for an echo detector submodule.
|
||||
class EchoDetector : public RefCountInterface {
|
||||
public:
|
||||
// (Re-)Initializes the submodule.
|
||||
virtual void Initialize(int capture_sample_rate_hz,
|
||||
int num_capture_channels,
|
||||
int render_sample_rate_hz,
|
||||
int num_render_channels) = 0;
|
||||
|
||||
// Analysis (not changing) of the first channel of the render signal.
|
||||
virtual void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) = 0;
|
||||
|
||||
// Analysis (not changing) of the capture signal.
|
||||
virtual void AnalyzeCaptureAudio(
|
||||
rtc::ArrayView<const float> capture_audio) = 0;
|
||||
|
||||
struct Metrics {
|
||||
absl::optional<double> echo_likelihood;
|
||||
absl::optional<double> echo_likelihood_recent_max;
|
||||
};
|
||||
|
||||
// Collect current metrics from the echo detector.
|
||||
virtual Metrics GetMetrics() const = 0;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // API_AUDIO_AUDIO_PROCESSING_H_
|
||||
|
|
@ -8,7 +8,7 @@
|
|||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
67
api/audio/audio_processing_statistics.h
Normal file
67
api/audio/audio_processing_statistics.h
Normal file
|
|
@ -0,0 +1,67 @@
|
|||
/*
|
||||
* Copyright 2017 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 API_AUDIO_AUDIO_PROCESSING_STATISTICS_H_
|
||||
#define API_AUDIO_AUDIO_PROCESSING_STATISTICS_H_
|
||||
|
||||
#include <stdint.h>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "rtc_base/system/rtc_export.h"
|
||||
|
||||
namespace webrtc {
|
||||
// This version of the stats uses Optionals, it will replace the regular
|
||||
// AudioProcessingStatistics struct.
|
||||
struct RTC_EXPORT AudioProcessingStats {
|
||||
AudioProcessingStats();
|
||||
AudioProcessingStats(const AudioProcessingStats& other);
|
||||
~AudioProcessingStats();
|
||||
|
||||
// Deprecated.
|
||||
// TODO(bugs.webrtc.org/11226): Remove.
|
||||
// True if voice is detected in the last capture frame, after processing.
|
||||
// It is conservative in flagging audio as speech, with low likelihood of
|
||||
// incorrectly flagging a frame as voice.
|
||||
// Only reported if voice detection is enabled in AudioProcessing::Config.
|
||||
absl::optional<bool> voice_detected;
|
||||
|
||||
// AEC Statistics.
|
||||
// ERL = 10log_10(P_far / P_echo)
|
||||
absl::optional<double> echo_return_loss;
|
||||
// ERLE = 10log_10(P_echo / P_out)
|
||||
absl::optional<double> echo_return_loss_enhancement;
|
||||
// Fraction of time that the AEC linear filter is divergent, in a 1-second
|
||||
// non-overlapped aggregation window.
|
||||
absl::optional<double> divergent_filter_fraction;
|
||||
|
||||
// The delay metrics consists of the delay median and standard deviation. It
|
||||
// also consists of the fraction of delay estimates that can make the echo
|
||||
// cancellation perform poorly. The values are aggregated until the first
|
||||
// call to `GetStatistics()` and afterwards aggregated and updated every
|
||||
// second. Note that if there are several clients pulling metrics from
|
||||
// `GetStatistics()` during a session the first call from any of them will
|
||||
// change to one second aggregation window for all.
|
||||
absl::optional<int32_t> delay_median_ms;
|
||||
absl::optional<int32_t> delay_standard_deviation_ms;
|
||||
|
||||
// Residual echo detector likelihood.
|
||||
absl::optional<double> residual_echo_likelihood;
|
||||
// Maximum residual echo likelihood from the last time period.
|
||||
absl::optional<double> residual_echo_likelihood_recent_max;
|
||||
|
||||
// The instantaneous delay estimate produced in the AEC. The unit is in
|
||||
// milliseconds and the value is the instantaneous value at the time of the
|
||||
// call to `GetStatistics()`.
|
||||
absl::optional<int32_t> delay_ms;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // API_AUDIO_AUDIO_PROCESSING_STATISTICS_H_
|
||||
|
|
@ -11,8 +11,8 @@
|
|||
#ifndef API_AUDIO_ECHO_DETECTOR_CREATOR_H_
|
||||
#define API_AUDIO_ECHO_DETECTOR_CREATOR_H_
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -13,6 +13,7 @@
|
|||
#include <memory>
|
||||
#include <utility>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/enable_media.h"
|
||||
#include "api/peer_connection_interface.h"
|
||||
#include "api/rtc_event_log/rtc_event_log_factory.h"
|
||||
|
|
@ -20,7 +21,6 @@
|
|||
#include "api/task_queue/default_task_queue_factory.h"
|
||||
#include "api/transport/field_trial_based_config.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/thread.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -14,6 +14,7 @@
|
|||
#include <memory>
|
||||
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
#include "api/field_trials_view.h"
|
||||
|
|
@ -34,7 +35,6 @@ namespace webrtc {
|
|||
|
||||
class AudioDeviceModule;
|
||||
class AudioFrameProcessor;
|
||||
class AudioProcessing;
|
||||
|
||||
// Create a new instance of PeerConnectionFactoryInterface with optional video
|
||||
// codec factories. These video factories represents all video codecs, i.e. no
|
||||
|
|
|
|||
|
|
@ -10,13 +10,13 @@
|
|||
|
||||
#include "api/enable_media_with_defaults.h"
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/enable_media.h"
|
||||
#include "api/task_queue/default_task_queue_factory.h"
|
||||
#include "api/video_codecs/builtin_video_decoder_factory.h"
|
||||
#include "api/video_codecs/builtin_video_encoder_factory.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -22,6 +22,7 @@
|
|||
#include <vector>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/audio_options.h"
|
||||
#include "api/ref_count.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
|
|
@ -30,7 +31,6 @@
|
|||
#include "api/video/video_sink_interface.h"
|
||||
#include "api/video/video_source_interface.h"
|
||||
#include "api/video_track_source_constraints.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "rtc_base/system/rtc_export.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -14,7 +14,7 @@
|
|||
#include <memory>
|
||||
#include <vector>
|
||||
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace test {
|
||||
|
|
|
|||
|
|
@ -32,7 +32,6 @@ rtc_source_set("media_configuration") {
|
|||
"../..:stats_observer_interface",
|
||||
"../..:track_id_stream_info_map",
|
||||
"../..:video_quality_analyzer_api",
|
||||
"../../../modules/audio_processing:api",
|
||||
"../../../rtc_base:checks",
|
||||
"../../../rtc_base:network",
|
||||
"../../../rtc_base:rtc_certificate_generator",
|
||||
|
|
@ -43,6 +42,7 @@ rtc_source_set("media_configuration") {
|
|||
"../../../test:video_test_support",
|
||||
"../../../test/pc/e2e/analyzer/video:video_dumping",
|
||||
"../../audio:audio_mixer_api",
|
||||
"../../audio:audio_processing",
|
||||
"../../rtc_event_log",
|
||||
"../../task_queue",
|
||||
"../../transport:network_control",
|
||||
|
|
@ -73,7 +73,6 @@ rtc_library("media_quality_test_params") {
|
|||
"../../../api/rtc_event_log",
|
||||
"../../../api/transport:network_control",
|
||||
"../../../api/video_codecs:video_codecs_api",
|
||||
"../../../modules/audio_processing:api",
|
||||
"../../../p2p:connection",
|
||||
"../../../p2p:port_allocator",
|
||||
"../../../p2p:rtc_p2p",
|
||||
|
|
@ -81,6 +80,7 @@ rtc_library("media_quality_test_params") {
|
|||
"../../../rtc_base:rtc_certificate_generator",
|
||||
"../../../rtc_base:ssl",
|
||||
"../../../rtc_base:threading",
|
||||
"../../audio:audio_processing",
|
||||
]
|
||||
}
|
||||
|
||||
|
|
@ -110,10 +110,10 @@ rtc_library("peer_configurer") {
|
|||
"../../../api/transport:bitrate_settings",
|
||||
"../../../api/transport:network_control",
|
||||
"../../../api/video_codecs:video_codecs_api",
|
||||
"../../../modules/audio_processing:api",
|
||||
"../../../rtc_base:checks",
|
||||
"../../../rtc_base:rtc_certificate_generator",
|
||||
"../../../rtc_base:ssl",
|
||||
"../../audio:audio_processing",
|
||||
]
|
||||
absl_deps = [
|
||||
"//third_party/abseil-cpp/absl/strings",
|
||||
|
|
|
|||
|
|
@ -25,6 +25,7 @@
|
|||
#include "absl/types/optional.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_options.h"
|
||||
#include "api/fec_controller.h"
|
||||
#include "api/function_view.h"
|
||||
|
|
@ -46,7 +47,6 @@
|
|||
#include "api/video_codecs/video_decoder_factory.h"
|
||||
#include "api/video_codecs/video_encoder.h"
|
||||
#include "api/video_codecs/video_encoder_factory.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/network.h"
|
||||
#include "rtc_base/rtc_certificate_generator.h"
|
||||
|
|
|
|||
|
|
@ -17,6 +17,7 @@
|
|||
|
||||
#include "api/async_dns_resolver.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/fec_controller.h"
|
||||
#include "api/field_trials_view.h"
|
||||
#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
|
||||
|
|
@ -24,7 +25,6 @@
|
|||
#include "api/transport/network_control.h"
|
||||
#include "api/video_codecs/video_decoder_factory.h"
|
||||
#include "api/video_codecs/video_encoder_factory.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
#include "rtc_base/network.h"
|
||||
#include "rtc_base/rtc_certificate_generator.h"
|
||||
|
|
|
|||
|
|
@ -20,6 +20,7 @@
|
|||
#include "absl/types/optional.h"
|
||||
#include "api/async_dns_resolver.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
#include "api/fec_controller.h"
|
||||
|
|
@ -38,7 +39,6 @@
|
|||
#include "api/transport/network_control.h"
|
||||
#include "api/video_codecs/video_decoder_factory.h"
|
||||
#include "api/video_codecs/video_encoder_factory.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/rtc_certificate_generator.h"
|
||||
#include "rtc_base/ssl_certificate.h"
|
||||
|
|
|
|||
|
|
@ -19,6 +19,7 @@
|
|||
#include "absl/types/variant.h"
|
||||
#include "api/async_dns_resolver.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
#include "api/fec_controller.h"
|
||||
|
|
@ -36,7 +37,6 @@
|
|||
#include "api/transport/network_control.h"
|
||||
#include "api/video_codecs/video_decoder_factory.h"
|
||||
#include "api/video_codecs/video_encoder_factory.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/rtc_certificate_generator.h"
|
||||
#include "rtc_base/ssl_certificate.h"
|
||||
|
||||
|
|
|
|||
|
|
@ -26,6 +26,7 @@
|
|||
#include "absl/types/optional.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/fec_controller.h"
|
||||
#include "api/function_view.h"
|
||||
#include "api/media_stream_interface.h"
|
||||
|
|
@ -50,7 +51,6 @@
|
|||
#include "api/video_codecs/video_encoder.h"
|
||||
#include "api/video_codecs/video_encoder_factory.h"
|
||||
#include "media/base/media_constants.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/network.h"
|
||||
#include "rtc_base/rtc_certificate_generator.h"
|
||||
|
|
|
|||
|
|
@ -41,8 +41,8 @@ rtc_library("voip_engine_factory") {
|
|||
"..:scoped_refptr",
|
||||
"../../audio/voip:voip_core",
|
||||
"../../modules/audio_device:audio_device_api",
|
||||
"../../modules/audio_processing:api",
|
||||
"../../rtc_base:logging",
|
||||
"../audio:audio_processing",
|
||||
"../audio_codecs:audio_codecs_api",
|
||||
"../task_queue",
|
||||
]
|
||||
|
|
|
|||
|
|
@ -13,13 +13,13 @@
|
|||
|
||||
#include <memory>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "api/task_queue/task_queue_factory.h"
|
||||
#include "api/voip/voip_engine.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -52,6 +52,7 @@ rtc_library("audio") {
|
|||
"../api/audio:audio_frame_api",
|
||||
"../api/audio:audio_frame_processor",
|
||||
"../api/audio:audio_mixer_api",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/crypto:frame_decryptor_interface",
|
||||
"../api/crypto:frame_encryptor_interface",
|
||||
|
|
@ -80,7 +81,6 @@ rtc_library("audio") {
|
|||
"../modules/audio_coding:red",
|
||||
"../modules/audio_device",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/audio_processing:audio_frame_proxies",
|
||||
"../modules/audio_processing:rms_level",
|
||||
"../modules/pacing",
|
||||
|
|
@ -171,6 +171,7 @@ if (rtc_include_tests) {
|
|||
"../api:mock_transformable_audio_frame",
|
||||
"../api:scoped_refptr",
|
||||
"../api/audio:audio_frame_api",
|
||||
"../api/audio:audio_processing_statistics",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/audio_codecs:builtin_audio_encoder_factory",
|
||||
"../api/audio_codecs/opus:audio_decoder_opus",
|
||||
|
|
@ -195,7 +196,6 @@ if (rtc_include_tests) {
|
|||
"../modules/audio_device:mock_audio_device",
|
||||
"../modules/audio_mixer:audio_mixer_impl",
|
||||
"../modules/audio_mixer:audio_mixer_test_utils",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../modules/audio_processing:mocks",
|
||||
"../modules/pacing",
|
||||
"../modules/rtp_rtcp:mock_rtp_rtcp",
|
||||
|
|
|
|||
|
|
@ -15,6 +15,7 @@
|
|||
#include <utility>
|
||||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/audio_encoder.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
#include "api/audio_codecs/audio_format.h"
|
||||
|
|
@ -34,7 +35,6 @@
|
|||
#include "media/base/media_channel.h"
|
||||
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
|
||||
#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/logging.h"
|
||||
|
|
|
|||
|
|
@ -16,6 +16,7 @@
|
|||
#include <utility>
|
||||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/task_queue/default_task_queue_factory.h"
|
||||
#include "api/test/mock_frame_encryptor.h"
|
||||
#include "audio/audio_state.h"
|
||||
|
|
@ -27,7 +28,6 @@
|
|||
#include "modules/audio_device/include/mock_audio_device.h"
|
||||
#include "modules/audio_mixer/audio_mixer_impl.h"
|
||||
#include "modules/audio_mixer/sine_wave_generator.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "modules/audio_processing/include/mock_audio_processing.h"
|
||||
#include "modules/rtp_rtcp/mocks/mock_network_link_rtcp_observer.h"
|
||||
#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
|
||||
|
|
|
|||
|
|
@ -15,11 +15,11 @@
|
|||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "common_audio/resampler/include/push_resampler.h"
|
||||
#include "modules/async_audio_processing/async_audio_processing.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/synchronization/mutex.h"
|
||||
#include "rtc_base/thread_annotations.h"
|
||||
|
||||
|
|
|
|||
|
|
@ -17,12 +17,12 @@ rtc_library("voip_core") {
|
|||
":audio_channel",
|
||||
"..:audio",
|
||||
"../../api:scoped_refptr",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../api/audio_codecs:audio_codecs_api",
|
||||
"../../api/task_queue",
|
||||
"../../api/voip:voip_api",
|
||||
"../../modules/audio_device:audio_device_api",
|
||||
"../../modules/audio_mixer:audio_mixer_impl",
|
||||
"../../modules/audio_processing:api",
|
||||
"../../rtc_base:criticalsection",
|
||||
"../../rtc_base:logging",
|
||||
"../../rtc_base/synchronization:mutex",
|
||||
|
|
|
|||
|
|
@ -17,6 +17,7 @@
|
|||
#include <unordered_map>
|
||||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
|
|
@ -32,7 +33,6 @@
|
|||
#include "audio/voip/audio_channel.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_mixer/audio_mixer_impl.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/synchronization/mutex.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -52,6 +52,8 @@ rtc_library("call_interfaces") {
|
|||
"../api/adaptation:resource_adaptation_api",
|
||||
"../api/audio:audio_frame_processor",
|
||||
"../api/audio:audio_mixer_api",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio:audio_processing_statistics",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/crypto:frame_encryptor_interface",
|
||||
"../api/crypto:options",
|
||||
|
|
@ -64,8 +66,6 @@ rtc_library("call_interfaces") {
|
|||
"../modules/async_audio_processing",
|
||||
"../modules/audio_device",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../modules/rtp_rtcp",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
"../rtc_base:audio_format_to_string",
|
||||
|
|
|
|||
|
|
@ -16,6 +16,7 @@
|
|||
#include <vector>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/audio_codecs/audio_codec_pair_id.h"
|
||||
#include "api/audio_codecs/audio_encoder.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
|
|
@ -29,7 +30,6 @@
|
|||
#include "api/scoped_refptr.h"
|
||||
#include "call/audio_sender.h"
|
||||
#include "call/rtp_config.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "modules/rtp_rtcp/include/report_block_data.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -11,10 +11,10 @@
|
|||
#define CALL_AUDIO_STATE_H_
|
||||
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "modules/async_audio_processing/async_audio_processing.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/ref_count.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -489,12 +489,12 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
|
|||
"../api:libjingle_peerconnection_api",
|
||||
"../api:scoped_refptr",
|
||||
"../api:sequence_checker",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio_codecs:builtin_audio_decoder_factory",
|
||||
"../api/audio_codecs:builtin_audio_encoder_factory",
|
||||
"../api/rtc_event_log:rtc_event_log_factory",
|
||||
"../api/task_queue:default_task_queue_factory",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../pc:libjingle_peerconnection",
|
||||
"../rtc_base/synchronization:mutex",
|
||||
"../sdk:base_objc",
|
||||
|
|
@ -690,6 +690,7 @@ if (is_linux || is_chromeos || is_win) {
|
|||
"../api:rtp_sender_interface",
|
||||
"../api:scoped_refptr",
|
||||
"../api/audio:audio_mixer_api",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/task_queue:pending_task_safety_flag",
|
||||
"../api/units:time_delta",
|
||||
|
|
@ -768,7 +769,6 @@ if (is_linux || is_chromeos || is_win) {
|
|||
"../media:rtc_audio_video",
|
||||
"../modules/audio_device",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/video_capture:video_capture_module",
|
||||
"../pc:libjingle_peerconnection",
|
||||
"../rtc_base:rtc_json",
|
||||
|
|
|
|||
|
|
@ -18,13 +18,13 @@
|
|||
#import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h"
|
||||
#import "sdk/objc/helpers/RTCCameraPreviewView.h"
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/enable_media.h"
|
||||
#include "api/peer_connection_interface.h"
|
||||
#include "api/rtc_event_log/rtc_event_log_factory.h"
|
||||
#include "api/task_queue/default_task_queue_factory.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "sdk/objc/native/api/video_capturer.h"
|
||||
#include "sdk/objc/native/api/video_decoder_factory.h"
|
||||
#include "sdk/objc/native/api/video_encoder_factory.h"
|
||||
|
|
|
|||
|
|
@ -20,6 +20,7 @@
|
|||
#include "absl/memory/memory.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
|
|
@ -41,7 +42,6 @@
|
|||
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
|
||||
#include "examples/peerconnection/client/defaults.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/video_capture/video_capture.h"
|
||||
#include "modules/video_capture/video_capture_factory.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
|
|
|
|||
|
|
@ -106,7 +106,6 @@ rtc_library("rtc_media_base") {
|
|||
"../common_video",
|
||||
"../modules/async_audio_processing",
|
||||
"../modules/audio_device",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
"../rtc_base:async_packet_socket",
|
||||
"../rtc_base:buffer",
|
||||
|
|
@ -329,6 +328,7 @@ rtc_source_set("media_channel") {
|
|||
"../api:rtp_parameters",
|
||||
"../api:rtp_sender_interface",
|
||||
"../api:scoped_refptr",
|
||||
"../api/audio:audio_processing_statistics",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/crypto:frame_decryptor_interface",
|
||||
"../api/crypto:frame_encryptor_interface",
|
||||
|
|
@ -344,7 +344,6 @@ rtc_source_set("media_channel") {
|
|||
"../api/video_codecs:video_codecs_api",
|
||||
"../call:video_stream_api",
|
||||
"../common_video",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
"../rtc_base:async_packet_socket",
|
||||
"../rtc_base:buffer",
|
||||
|
|
@ -581,6 +580,8 @@ rtc_library("rtc_audio_video") {
|
|||
"../api/audio:audio_frame_api",
|
||||
"../api/audio:audio_frame_processor",
|
||||
"../api/audio:audio_mixer_api",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio:audio_processing_statistics",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/crypto:frame_decryptor_interface",
|
||||
"../api/crypto:frame_encryptor_interface",
|
||||
|
|
@ -615,8 +616,6 @@ rtc_library("rtc_audio_video") {
|
|||
"../modules/audio_device",
|
||||
"../modules/audio_device:audio_device_impl",
|
||||
"../modules/audio_mixer:audio_mixer_impl",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../modules/audio_processing/aec_dump",
|
||||
"../modules/audio_processing/agc:gain_control_interface",
|
||||
"../modules/rtp_rtcp",
|
||||
|
|
@ -807,6 +806,7 @@ if (rtc_include_tests) {
|
|||
"../api:fec_controller_api",
|
||||
"../api:rtp_parameters",
|
||||
"../api:scoped_refptr",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/environment",
|
||||
"../api/task_queue",
|
||||
"../api/task_queue:pending_task_safety_flag",
|
||||
|
|
@ -821,7 +821,6 @@ if (rtc_include_tests) {
|
|||
"../call:video_stream_api",
|
||||
"../common_video",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
"../modules/video_coding:video_codec_interface",
|
||||
"../modules/video_coding:video_coding_utility",
|
||||
|
|
@ -955,7 +954,6 @@ if (rtc_include_tests) {
|
|||
"../modules/audio_device:mock_audio_device",
|
||||
"../modules/audio_mixer:audio_mixer_impl",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/audio_processing:mocks",
|
||||
"../modules/rtp_rtcp",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
|
|
|
|||
|
|
@ -23,6 +23,7 @@
|
|||
|
||||
#include "absl/algorithm/container.h"
|
||||
#include "absl/functional/any_invocable.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/call/audio_sink.h"
|
||||
#include "api/media_types.h"
|
||||
#include "media/base/audio_source.h"
|
||||
|
|
@ -32,7 +33,6 @@
|
|||
#include "media/base/rtp_utils.h"
|
||||
#include "media/base/stream_params.h"
|
||||
#include "media/engine/webrtc_video_engine.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
|
||||
#include "rtc_base/copy_on_write_buffer.h"
|
||||
#include "rtc_base/network_route.h"
|
||||
|
|
|
|||
|
|
@ -19,6 +19,7 @@
|
|||
#include <vector>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/audio_codecs/audio_encoder.h"
|
||||
#include "api/audio_options.h"
|
||||
#include "api/call/audio_sink.h"
|
||||
|
|
@ -45,7 +46,6 @@
|
|||
#include "media/base/codec.h"
|
||||
#include "media/base/media_constants.h"
|
||||
#include "media/base/stream_params.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "modules/rtp_rtcp/include/report_block_data.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
|
||||
#include "rtc_base/async_packet_socket.h"
|
||||
|
|
|
|||
|
|
@ -32,7 +32,6 @@
|
|||
namespace webrtc {
|
||||
class AudioDeviceModule;
|
||||
class AudioMixer;
|
||||
class AudioProcessing;
|
||||
class Call;
|
||||
} // namespace webrtc
|
||||
|
||||
|
|
|
|||
|
|
@ -29,6 +29,8 @@
|
|||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_frame.h"
|
||||
#include "api/audio/audio_frame_processor.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/audio_codecs/audio_codec_pair_id.h"
|
||||
#include "api/audio_codecs/audio_encoder.h"
|
||||
#include "api/call/audio_sink.h"
|
||||
|
|
@ -57,8 +59,6 @@
|
|||
#include "media/engine/webrtc_media_engine.h"
|
||||
#include "modules/async_audio_processing/async_audio_processing.h"
|
||||
#include "modules/audio_mixer/audio_mixer_impl.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "modules/rtp_rtcp/include/report_block_data.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
|
||||
|
|
|
|||
|
|
@ -26,6 +26,7 @@
|
|||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_frame_processor.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/audio_codec_pair_id.h"
|
||||
#include "api/audio_codecs/audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
|
|
@ -59,7 +60,6 @@
|
|||
#include "media/base/stream_params.h"
|
||||
#include "modules/async_audio_processing/async_audio_processing.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
|
||||
#include "rtc_base/buffer.h"
|
||||
|
|
|
|||
|
|
@ -43,6 +43,8 @@ rtc_library("audio_mixer_impl") {
|
|||
"../../api:scoped_refptr",
|
||||
"../../api/audio:audio_frame_api",
|
||||
"../../api/audio:audio_mixer_api",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../audio/utility:audio_frame_operations",
|
||||
"../../common_audio",
|
||||
"../../rtc_base:checks",
|
||||
|
|
@ -55,7 +57,6 @@ rtc_library("audio_mixer_impl") {
|
|||
"../../rtc_base/synchronization:mutex",
|
||||
"../../system_wrappers",
|
||||
"../../system_wrappers:metrics",
|
||||
"../audio_processing:api",
|
||||
"../audio_processing:apm_logging",
|
||||
"../audio_processing:audio_frame_view",
|
||||
"../audio_processing/agc2:fixed_digital",
|
||||
|
|
|
|||
|
|
@ -13,7 +13,7 @@
|
|||
#include <algorithm>
|
||||
#include <iterator>
|
||||
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "rtc_base/checks.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -20,13 +20,13 @@
|
|||
#include <vector>
|
||||
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/rtp_packet_info.h"
|
||||
#include "api/rtp_packet_infos.h"
|
||||
#include "common_audio/include/audio_util.h"
|
||||
#include "modules/audio_mixer/audio_frame_manipulator.h"
|
||||
#include "modules/audio_mixer/audio_mixer_impl.h"
|
||||
#include "modules/audio_processing/include/audio_frame_view.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
||||
#include "rtc_base/arraysize.h"
|
||||
#include "rtc_base/checks.h"
|
||||
|
|
|
|||
|
|
@ -21,33 +21,8 @@ config("apm_debug_dump") {
|
|||
|
||||
rtc_library("api") {
|
||||
visibility = [ "*" ]
|
||||
sources = [
|
||||
"include/audio_processing.cc",
|
||||
"include/audio_processing.h",
|
||||
]
|
||||
deps = [
|
||||
":audio_frame_view",
|
||||
":audio_processing_statistics",
|
||||
"../../api:array_view",
|
||||
"../../api:ref_count",
|
||||
"../../api:scoped_refptr",
|
||||
"../../api/audio:aec3_config",
|
||||
"../../api/audio:audio_frame_api",
|
||||
"../../api/audio:echo_control",
|
||||
"../../api/task_queue",
|
||||
"../../rtc_base:macromagic",
|
||||
"../../rtc_base:refcount",
|
||||
"../../rtc_base:stringutils",
|
||||
"../../rtc_base/system:arch",
|
||||
"../../rtc_base/system:file_wrapper",
|
||||
"../../rtc_base/system:rtc_export",
|
||||
"agc:gain_control_interface",
|
||||
]
|
||||
absl_deps = [
|
||||
"//third_party/abseil-cpp/absl/base:nullability",
|
||||
"//third_party/abseil-cpp/absl/strings",
|
||||
"//third_party/abseil-cpp/absl/types:optional",
|
||||
]
|
||||
sources = [ "include/audio_processing.h" ]
|
||||
deps = [ "../../api/audio:audio_processing" ]
|
||||
}
|
||||
|
||||
rtc_library("audio_frame_proxies") {
|
||||
|
|
@ -57,9 +32,9 @@ rtc_library("audio_frame_proxies") {
|
|||
"include/audio_frame_proxies.h",
|
||||
]
|
||||
deps = [
|
||||
":api",
|
||||
":audio_frame_view",
|
||||
"../../api/audio:audio_frame_api",
|
||||
"../../api/audio:audio_processing",
|
||||
]
|
||||
}
|
||||
|
||||
|
|
@ -80,8 +55,8 @@ rtc_library("audio_buffer") {
|
|||
defines = []
|
||||
|
||||
deps = [
|
||||
":api",
|
||||
"../../api:array_view",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../common_audio",
|
||||
"../../common_audio:common_audio_c",
|
||||
"../../rtc_base:checks",
|
||||
|
|
@ -114,8 +89,8 @@ rtc_source_set("aec_dump_interface") {
|
|||
]
|
||||
|
||||
deps = [
|
||||
":api",
|
||||
":audio_frame_view",
|
||||
"../../api/audio:audio_processing",
|
||||
]
|
||||
absl_deps = [
|
||||
"//third_party/abseil-cpp/absl/base:core_headers",
|
||||
|
|
@ -132,10 +107,10 @@ rtc_library("gain_controller2") {
|
|||
defines = []
|
||||
deps = [
|
||||
":aec_dump_interface",
|
||||
":api",
|
||||
":apm_logging",
|
||||
":audio_buffer",
|
||||
":audio_frame_view",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../common_audio",
|
||||
"../../rtc_base:checks",
|
||||
"../../rtc_base:logging",
|
||||
|
|
@ -171,12 +146,10 @@ rtc_library("audio_processing") {
|
|||
defines = []
|
||||
deps = [
|
||||
":aec_dump_interface",
|
||||
":api",
|
||||
":apm_logging",
|
||||
":audio_buffer",
|
||||
":audio_frame_proxies",
|
||||
":audio_frame_view",
|
||||
":audio_processing_statistics",
|
||||
":gain_controller2",
|
||||
":high_pass_filter",
|
||||
":optionally_built_submodule_creators",
|
||||
|
|
@ -186,6 +159,8 @@ rtc_library("audio_processing") {
|
|||
"../../api:make_ref_counted",
|
||||
"../../api/audio:aec3_config",
|
||||
"../../api/audio:audio_frame_api",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../api/audio:audio_processing_statistics",
|
||||
"../../api/audio:echo_control",
|
||||
"../../api/task_queue",
|
||||
"../../audio/utility:audio_frame_operations",
|
||||
|
|
@ -255,9 +230,9 @@ rtc_library("residual_echo_detector") {
|
|||
"residual_echo_detector.h",
|
||||
]
|
||||
deps = [
|
||||
":api",
|
||||
":apm_logging",
|
||||
"../../api:array_view",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../rtc_base:checks",
|
||||
"../../rtc_base:logging",
|
||||
"../../system_wrappers:metrics",
|
||||
|
|
@ -291,12 +266,8 @@ rtc_source_set("rms_level") {
|
|||
|
||||
rtc_library("audio_processing_statistics") {
|
||||
visibility = [ "*" ]
|
||||
sources = [
|
||||
"include/audio_processing_statistics.cc",
|
||||
"include/audio_processing_statistics.h",
|
||||
]
|
||||
deps = [ "../../rtc_base/system:rtc_export" ]
|
||||
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
|
||||
sources = [ "include/audio_processing_statistics.h" ]
|
||||
deps = [ "../../api/audio:audio_processing_statistics" ]
|
||||
}
|
||||
|
||||
rtc_source_set("audio_frame_view") {
|
||||
|
|
@ -337,10 +308,10 @@ if (rtc_include_tests) {
|
|||
sources = [ "include/mock_audio_processing.h" ]
|
||||
deps = [
|
||||
":aec_dump_interface",
|
||||
":api",
|
||||
":audio_buffer",
|
||||
":audio_processing",
|
||||
":audio_processing_statistics",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../api/audio:audio_processing_statistics",
|
||||
"../../api/task_queue",
|
||||
"../../test:test_support",
|
||||
]
|
||||
|
|
@ -386,7 +357,6 @@ if (rtc_include_tests) {
|
|||
deps = [
|
||||
":aec3_config_json",
|
||||
":analog_mic_simulation",
|
||||
":api",
|
||||
":apm_logging",
|
||||
":audio_buffer",
|
||||
":audio_frame_view",
|
||||
|
|
@ -400,6 +370,7 @@ if (rtc_include_tests) {
|
|||
"../../api:scoped_refptr",
|
||||
"../../api/audio:aec3_config",
|
||||
"../../api/audio:aec3_factory",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../api/audio:echo_detector_creator",
|
||||
"../../common_audio",
|
||||
"../../common_audio:common_audio_c",
|
||||
|
|
@ -568,7 +539,6 @@ if (rtc_include_tests) {
|
|||
deps = [
|
||||
":aec3_config_json",
|
||||
":analog_mic_simulation",
|
||||
":api",
|
||||
":apm_logging",
|
||||
":audio_processing",
|
||||
":audioproc_debug_proto",
|
||||
|
|
@ -576,6 +546,7 @@ if (rtc_include_tests) {
|
|||
":audioproc_test_utils",
|
||||
":runtime_settings_protobuf_utils",
|
||||
"../../api/audio:aec3_factory",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../api/audio:echo_detector_creator",
|
||||
"../../common_audio",
|
||||
"../../rtc_base:checks",
|
||||
|
|
@ -631,9 +602,9 @@ if (rtc_include_tests) {
|
|||
]
|
||||
|
||||
deps = [
|
||||
":api",
|
||||
":audioproc_debug_proto",
|
||||
":audioproc_protobuf_utils",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../rtc_base:checks",
|
||||
]
|
||||
}
|
||||
|
|
@ -661,11 +632,11 @@ rtc_library("audioproc_test_utils") {
|
|||
configs += [ ":apm_debug_dump" ]
|
||||
|
||||
deps = [
|
||||
":api",
|
||||
":audio_buffer",
|
||||
":audio_processing",
|
||||
"../../api:array_view",
|
||||
"../../api/audio:audio_frame_api",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../common_audio",
|
||||
"../../rtc_base:checks",
|
||||
"../../rtc_base:random",
|
||||
|
|
|
|||
|
|
@ -47,9 +47,9 @@ if (rtc_include_tests) {
|
|||
|
||||
deps = [
|
||||
":mock_aec_dump",
|
||||
"..:api",
|
||||
"..:audioproc_test_utils",
|
||||
"../",
|
||||
"../../../api/audio:audio_processing",
|
||||
"//testing/gtest",
|
||||
]
|
||||
}
|
||||
|
|
|
|||
|
|
@ -12,9 +12,9 @@
|
|||
#include <memory>
|
||||
#include <utility>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
|
||||
#include "modules/audio_processing/audio_processing_impl.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
|
||||
|
||||
using ::testing::_;
|
||||
|
|
|
|||
|
|
@ -21,11 +21,11 @@ rtc_library("agc") {
|
|||
deps = [
|
||||
":gain_control_interface",
|
||||
":level_estimation",
|
||||
"..:api",
|
||||
"..:apm_logging",
|
||||
"..:audio_buffer",
|
||||
"..:audio_frame_view",
|
||||
"../../../api:array_view",
|
||||
"../../../api/audio:audio_processing",
|
||||
"../../../common_audio",
|
||||
"../../../common_audio:common_audio_c",
|
||||
"../../../rtc_base:checks",
|
||||
|
|
|
|||
|
|
@ -16,10 +16,10 @@
|
|||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/agc/agc.h"
|
||||
#include "modules/audio_processing/agc2/clipping_predictor.h"
|
||||
#include "modules/audio_processing/audio_buffer.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
||||
#include "rtc_base/gtest_prod_util.h"
|
||||
|
||||
|
|
|
|||
|
|
@ -23,9 +23,9 @@ rtc_library("speech_level_estimator") {
|
|||
|
||||
deps = [
|
||||
":common",
|
||||
"..:api",
|
||||
"..:apm_logging",
|
||||
"../../../api:array_view",
|
||||
"../../../api/audio:audio_processing",
|
||||
"../../../rtc_base:checks",
|
||||
"../../../rtc_base:logging",
|
||||
"../../../rtc_base:safe_minmax",
|
||||
|
|
@ -48,9 +48,9 @@ rtc_library("adaptive_digital_gain_controller") {
|
|||
deps = [
|
||||
":common",
|
||||
":gain_applier",
|
||||
"..:api",
|
||||
"..:apm_logging",
|
||||
"..:audio_frame_view",
|
||||
"../../../api/audio:audio_processing",
|
||||
"../../../common_audio",
|
||||
"../../../rtc_base:checks",
|
||||
"../../../rtc_base:logging",
|
||||
|
|
@ -112,8 +112,8 @@ rtc_library("clipping_predictor") {
|
|||
|
||||
deps = [
|
||||
":gain_map",
|
||||
"..:api",
|
||||
"..:audio_frame_view",
|
||||
"../../../api/audio:audio_processing",
|
||||
"../../../common_audio",
|
||||
"../../../rtc_base:checks",
|
||||
"../../../rtc_base:logging",
|
||||
|
|
@ -209,10 +209,10 @@ rtc_library("input_volume_controller") {
|
|||
":clipping_predictor",
|
||||
":gain_map",
|
||||
":input_volume_stats_reporter",
|
||||
"..:api",
|
||||
"..:audio_buffer",
|
||||
"..:audio_frame_view",
|
||||
"../../../api:array_view",
|
||||
"../../../api/audio:audio_processing",
|
||||
"../../../rtc_base:checks",
|
||||
"../../../rtc_base:checks",
|
||||
"../../../rtc_base:gtest_prod",
|
||||
|
|
@ -303,8 +303,8 @@ rtc_library("speech_level_estimator_unittest") {
|
|||
deps = [
|
||||
":common",
|
||||
":speech_level_estimator",
|
||||
"..:api",
|
||||
"..:apm_logging",
|
||||
"../../../api/audio:audio_processing",
|
||||
"../../../rtc_base:gunit_helpers",
|
||||
"../../../test:test_support",
|
||||
]
|
||||
|
|
@ -320,9 +320,9 @@ rtc_library("adaptive_digital_gain_controller_unittest") {
|
|||
":adaptive_digital_gain_controller",
|
||||
":common",
|
||||
":test_utils",
|
||||
"..:api",
|
||||
"..:apm_logging",
|
||||
"..:audio_frame_view",
|
||||
"../../../api/audio:audio_processing",
|
||||
"../../../common_audio",
|
||||
"../../../rtc_base:gunit_helpers",
|
||||
"../../../test:test_support",
|
||||
|
|
@ -413,7 +413,6 @@ rtc_library("input_volume_controller_unittests") {
|
|||
":clipping_predictor",
|
||||
":gain_map",
|
||||
":input_volume_controller",
|
||||
"..:api",
|
||||
"../../../api:array_view",
|
||||
"../../../rtc_base:checks",
|
||||
"../../../rtc_base:random",
|
||||
|
|
|
|||
|
|
@ -13,9 +13,9 @@
|
|||
|
||||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/agc2/gain_applier.h"
|
||||
#include "modules/audio_processing/include/audio_frame_view.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -13,10 +13,10 @@
|
|||
#include <algorithm>
|
||||
#include <memory>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "common_audio/include/audio_util.h"
|
||||
#include "modules/audio_processing/agc2/agc2_common.h"
|
||||
#include "modules/audio_processing/agc2/vector_float_frame.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
||||
#include "rtc_base/gunit.h"
|
||||
|
||||
|
|
|
|||
|
|
@ -15,8 +15,8 @@
|
|||
#include <vector>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/include/audio_frame_view.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -16,9 +16,9 @@
|
|||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/agc2/clipping_predictor.h"
|
||||
#include "modules/audio_processing/audio_buffer.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/gtest_prod_util.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -15,8 +15,8 @@
|
|||
|
||||
#include <type_traits>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/agc2/agc2_common.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
class ApmDataDumper;
|
||||
|
|
|
|||
|
|
@ -12,8 +12,8 @@
|
|||
|
||||
#include <memory>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/agc2/agc2_common.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
||||
#include "rtc_base/gunit.h"
|
||||
|
||||
|
|
|
|||
|
|
@ -17,8 +17,8 @@
|
|||
#include <memory>
|
||||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "common_audio/channel_buffer.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -10,9 +10,9 @@
|
|||
|
||||
#include <memory>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/make_ref_counted.h"
|
||||
#include "modules/audio_processing/audio_processing_impl.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -23,6 +23,8 @@
|
|||
#include "absl/strings/string_view.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/function_view.h"
|
||||
#include "api/task_queue/task_queue_base.h"
|
||||
#include "modules/audio_processing/aec3/echo_canceller3.h"
|
||||
|
|
@ -37,8 +39,6 @@
|
|||
#include "modules/audio_processing/high_pass_filter.h"
|
||||
#include "modules/audio_processing/include/aec_dump.h"
|
||||
#include "modules/audio_processing/include/audio_frame_proxies.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "modules/audio_processing/ns/noise_suppressor.h"
|
||||
#include "modules/audio_processing/optionally_built_submodule_creators.h"
|
||||
#include "modules/audio_processing/render_queue_item_verifier.h"
|
||||
|
|
|
|||
|
|
@ -16,9 +16,9 @@
|
|||
#include <tuple>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/make_ref_counted.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/optionally_built_submodule_creators.h"
|
||||
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
|
||||
#include "modules/audio_processing/test/echo_canceller_test_tools.h"
|
||||
|
|
|
|||
|
|
@ -7,7 +7,7 @@
|
|||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
|
||||
#include <math.h>
|
||||
#include <stdio.h>
|
||||
|
|
|
|||
|
|
@ -14,9 +14,9 @@
|
|||
|
||||
#include <cstdint>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/aecm/echo_control_mobile.h"
|
||||
#include "modules/audio_processing/audio_buffer.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/checks.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -11,8 +11,8 @@
|
|||
#include <array>
|
||||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/echo_control_mobile_impl.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "test/gtest.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -13,9 +13,9 @@
|
|||
#include <cstdint>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/agc/legacy/gain_control.h"
|
||||
#include "modules/audio_processing/audio_buffer.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/logging.h"
|
||||
|
|
|
|||
|
|
@ -15,6 +15,7 @@
|
|||
#include <memory>
|
||||
#include <string>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
|
||||
#include "modules/audio_processing/agc2/cpu_features.h"
|
||||
#include "modules/audio_processing/agc2/gain_applier.h"
|
||||
|
|
@ -24,7 +25,6 @@
|
|||
#include "modules/audio_processing/agc2/saturation_protector.h"
|
||||
#include "modules/audio_processing/agc2/speech_level_estimator.h"
|
||||
#include "modules/audio_processing/agc2/vad_wrapper.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -17,8 +17,8 @@
|
|||
|
||||
#include "absl/base/attributes.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/include/audio_frame_view.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -11,7 +11,7 @@
|
|||
#include "modules/audio_processing/include/audio_frame_proxies.h"
|
||||
|
||||
#include "api/audio/audio_frame.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -11,931 +11,8 @@
|
|||
#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
|
||||
#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
|
||||
|
||||
// MSVC++ requires this to be set before any other includes to get M_PI.
|
||||
#ifndef _USE_MATH_DEFINES
|
||||
#define _USE_MATH_DEFINES
|
||||
#endif
|
||||
|
||||
#include <math.h>
|
||||
#include <stddef.h> // size_t
|
||||
#include <stdio.h> // FILE
|
||||
#include <string.h>
|
||||
|
||||
#include <vector>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/echo_canceller3_config.h"
|
||||
#include "api/audio/echo_control.h"
|
||||
#include "api/ref_count.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "api/task_queue/task_queue_base.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "rtc_base/arraysize.h"
|
||||
#include "rtc_base/system/file_wrapper.h"
|
||||
#include "rtc_base/system/rtc_export.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
class AecDump;
|
||||
class AudioBuffer;
|
||||
|
||||
class StreamConfig;
|
||||
class ProcessingConfig;
|
||||
|
||||
class EchoDetector;
|
||||
class CustomAudioAnalyzer;
|
||||
class CustomProcessing;
|
||||
|
||||
// The Audio Processing Module (APM) provides a collection of voice processing
|
||||
// components designed for real-time communications software.
|
||||
//
|
||||
// APM operates on two audio streams on a frame-by-frame basis. Frames of the
|
||||
// primary stream, on which all processing is applied, are passed to
|
||||
// `ProcessStream()`. Frames of the reverse direction stream are passed to
|
||||
// `ProcessReverseStream()`. On the client-side, this will typically be the
|
||||
// near-end (capture) and far-end (render) streams, respectively. APM should be
|
||||
// placed in the signal chain as close to the audio hardware abstraction layer
|
||||
// (HAL) as possible.
|
||||
//
|
||||
// On the server-side, the reverse stream will normally not be used, with
|
||||
// processing occurring on each incoming stream.
|
||||
//
|
||||
// Component interfaces follow a similar pattern and are accessed through
|
||||
// corresponding getters in APM. All components are disabled at create-time,
|
||||
// with default settings that are recommended for most situations. New settings
|
||||
// can be applied without enabling a component. Enabling a component triggers
|
||||
// memory allocation and initialization to allow it to start processing the
|
||||
// streams.
|
||||
//
|
||||
// Thread safety is provided with the following assumptions to reduce locking
|
||||
// overhead:
|
||||
// 1. The stream getters and setters are called from the same thread as
|
||||
// ProcessStream(). More precisely, stream functions are never called
|
||||
// concurrently with ProcessStream().
|
||||
// 2. Parameter getters are never called concurrently with the corresponding
|
||||
// setter.
|
||||
//
|
||||
// APM accepts only linear PCM audio data in chunks of ~10 ms (see
|
||||
// AudioProcessing::GetFrameSize() for details) and sample rates ranging from
|
||||
// 8000 Hz to 384000 Hz. The int16 interfaces use interleaved data, while the
|
||||
// float interfaces use deinterleaved data.
|
||||
//
|
||||
// Usage example, omitting error checking:
|
||||
// rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
|
||||
//
|
||||
// AudioProcessing::Config config;
|
||||
// config.echo_canceller.enabled = true;
|
||||
// config.echo_canceller.mobile_mode = false;
|
||||
//
|
||||
// config.gain_controller1.enabled = true;
|
||||
// config.gain_controller1.mode =
|
||||
// AudioProcessing::Config::GainController1::kAdaptiveAnalog;
|
||||
// config.gain_controller1.analog_level_minimum = 0;
|
||||
// config.gain_controller1.analog_level_maximum = 255;
|
||||
//
|
||||
// config.gain_controller2.enabled = true;
|
||||
//
|
||||
// config.high_pass_filter.enabled = true;
|
||||
//
|
||||
// apm->ApplyConfig(config)
|
||||
//
|
||||
// // Start a voice call...
|
||||
//
|
||||
// // ... Render frame arrives bound for the audio HAL ...
|
||||
// apm->ProcessReverseStream(render_frame);
|
||||
//
|
||||
// // ... Capture frame arrives from the audio HAL ...
|
||||
// // Call required set_stream_ functions.
|
||||
// apm->set_stream_delay_ms(delay_ms);
|
||||
// apm->set_stream_analog_level(analog_level);
|
||||
//
|
||||
// apm->ProcessStream(capture_frame);
|
||||
//
|
||||
// // Call required stream_ functions.
|
||||
// analog_level = apm->recommended_stream_analog_level();
|
||||
// has_voice = apm->stream_has_voice();
|
||||
//
|
||||
// // Repeat render and capture processing for the duration of the call...
|
||||
// // Start a new call...
|
||||
// apm->Initialize();
|
||||
//
|
||||
// // Close the application...
|
||||
// apm.reset();
|
||||
//
|
||||
class RTC_EXPORT AudioProcessing : public RefCountInterface {
|
||||
public:
|
||||
// The struct below constitutes the new parameter scheme for the audio
|
||||
// processing. It is being introduced gradually and until it is fully
|
||||
// introduced, it is prone to change.
|
||||
// TODO(peah): Remove this comment once the new config scheme is fully rolled
|
||||
// out.
|
||||
//
|
||||
// The parameters and behavior of the audio processing module are controlled
|
||||
// by changing the default values in the AudioProcessing::Config struct.
|
||||
// The config is applied by passing the struct to the ApplyConfig method.
|
||||
//
|
||||
// This config is intended to be used during setup, and to enable/disable
|
||||
// top-level processing effects. Use during processing may cause undesired
|
||||
// submodule resets, affecting the audio quality. Use the RuntimeSetting
|
||||
// construct for runtime configuration.
|
||||
struct RTC_EXPORT Config {
|
||||
// Sets the properties of the audio processing pipeline.
|
||||
struct RTC_EXPORT Pipeline {
|
||||
// Ways to downmix a multi-channel track to mono.
|
||||
enum class DownmixMethod {
|
||||
kAverageChannels, // Average across channels.
|
||||
kUseFirstChannel // Use the first channel.
|
||||
};
|
||||
|
||||
// Maximum allowed processing rate used internally. May only be set to
|
||||
// 32000 or 48000 and any differing values will be treated as 48000.
|
||||
int maximum_internal_processing_rate = 48000;
|
||||
// Allow multi-channel processing of render audio.
|
||||
bool multi_channel_render = false;
|
||||
// Allow multi-channel processing of capture audio when AEC3 is active
|
||||
// or a custom AEC is injected..
|
||||
bool multi_channel_capture = false;
|
||||
// Indicates how to downmix multi-channel capture audio to mono (when
|
||||
// needed).
|
||||
DownmixMethod capture_downmix_method = DownmixMethod::kAverageChannels;
|
||||
} pipeline;
|
||||
|
||||
// Enabled the pre-amplifier. It amplifies the capture signal
|
||||
// before any other processing is done.
|
||||
// TODO(webrtc:5298): Deprecate and use the pre-gain functionality in
|
||||
// capture_level_adjustment instead.
|
||||
struct PreAmplifier {
|
||||
bool enabled = false;
|
||||
float fixed_gain_factor = 1.0f;
|
||||
} pre_amplifier;
|
||||
|
||||
// Functionality for general level adjustment in the capture pipeline. This
|
||||
// should not be used together with the legacy PreAmplifier functionality.
|
||||
struct CaptureLevelAdjustment {
|
||||
bool operator==(const CaptureLevelAdjustment& rhs) const;
|
||||
bool operator!=(const CaptureLevelAdjustment& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
bool enabled = false;
|
||||
// The `pre_gain_factor` scales the signal before any processing is done.
|
||||
float pre_gain_factor = 1.0f;
|
||||
// The `post_gain_factor` scales the signal after all processing is done.
|
||||
float post_gain_factor = 1.0f;
|
||||
struct AnalogMicGainEmulation {
|
||||
bool operator==(const AnalogMicGainEmulation& rhs) const;
|
||||
bool operator!=(const AnalogMicGainEmulation& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
bool enabled = false;
|
||||
// Initial analog gain level to use for the emulated analog gain. Must
|
||||
// be in the range [0...255].
|
||||
int initial_level = 255;
|
||||
} analog_mic_gain_emulation;
|
||||
} capture_level_adjustment;
|
||||
|
||||
struct HighPassFilter {
|
||||
bool enabled = false;
|
||||
bool apply_in_full_band = true;
|
||||
} high_pass_filter;
|
||||
|
||||
struct EchoCanceller {
|
||||
bool enabled = false;
|
||||
bool mobile_mode = false;
|
||||
bool export_linear_aec_output = false;
|
||||
// Enforce the highpass filter to be on (has no effect for the mobile
|
||||
// mode).
|
||||
bool enforce_high_pass_filtering = true;
|
||||
} echo_canceller;
|
||||
|
||||
// Enables background noise suppression.
|
||||
struct NoiseSuppression {
|
||||
bool enabled = false;
|
||||
enum Level { kLow, kModerate, kHigh, kVeryHigh };
|
||||
Level level = kModerate;
|
||||
bool analyze_linear_aec_output_when_available = false;
|
||||
} noise_suppression;
|
||||
|
||||
// Enables transient suppression.
|
||||
struct TransientSuppression {
|
||||
bool enabled = false;
|
||||
} transient_suppression;
|
||||
|
||||
// Enables automatic gain control (AGC) functionality.
|
||||
// The automatic gain control (AGC) component brings the signal to an
|
||||
// appropriate range. This is done by applying a digital gain directly and,
|
||||
// in the analog mode, prescribing an analog gain to be applied at the audio
|
||||
// HAL.
|
||||
// Recommended to be enabled on the client-side.
|
||||
struct RTC_EXPORT GainController1 {
|
||||
bool operator==(const GainController1& rhs) const;
|
||||
bool operator!=(const GainController1& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
|
||||
bool enabled = false;
|
||||
enum Mode {
|
||||
// Adaptive mode intended for use if an analog volume control is
|
||||
// available on the capture device. It will require the user to provide
|
||||
// coupling between the OS mixer controls and AGC through the
|
||||
// stream_analog_level() functions.
|
||||
// It consists of an analog gain prescription for the audio device and a
|
||||
// digital compression stage.
|
||||
kAdaptiveAnalog,
|
||||
// Adaptive mode intended for situations in which an analog volume
|
||||
// control is unavailable. It operates in a similar fashion to the
|
||||
// adaptive analog mode, but with scaling instead applied in the digital
|
||||
// domain. As with the analog mode, it additionally uses a digital
|
||||
// compression stage.
|
||||
kAdaptiveDigital,
|
||||
// Fixed mode which enables only the digital compression stage also used
|
||||
// by the two adaptive modes.
|
||||
// It is distinguished from the adaptive modes by considering only a
|
||||
// short time-window of the input signal. It applies a fixed gain
|
||||
// through most of the input level range, and compresses (gradually
|
||||
// reduces gain with increasing level) the input signal at higher
|
||||
// levels. This mode is preferred on embedded devices where the capture
|
||||
// signal level is predictable, so that a known gain can be applied.
|
||||
kFixedDigital
|
||||
};
|
||||
Mode mode = kAdaptiveAnalog;
|
||||
// Sets the target peak level (or envelope) of the AGC in dBFs (decibels
|
||||
// from digital full-scale). The convention is to use positive values. For
|
||||
// instance, passing in a value of 3 corresponds to -3 dBFs, or a target
|
||||
// level 3 dB below full-scale. Limited to [0, 31].
|
||||
int target_level_dbfs = 3;
|
||||
// Sets the maximum gain the digital compression stage may apply, in dB. A
|
||||
// higher number corresponds to greater compression, while a value of 0
|
||||
// will leave the signal uncompressed. Limited to [0, 90].
|
||||
// For updates after APM setup, use a RuntimeSetting instead.
|
||||
int compression_gain_db = 9;
|
||||
// When enabled, the compression stage will hard limit the signal to the
|
||||
// target level. Otherwise, the signal will be compressed but not limited
|
||||
// above the target level.
|
||||
bool enable_limiter = true;
|
||||
|
||||
// Enables the analog gain controller functionality.
|
||||
struct AnalogGainController {
|
||||
bool enabled = true;
|
||||
// TODO(bugs.webrtc.org/7494): Deprecated. Stop using and remove.
|
||||
int startup_min_volume = 0;
|
||||
// Lowest analog microphone level that will be applied in response to
|
||||
// clipping.
|
||||
int clipped_level_min = 70;
|
||||
// If true, an adaptive digital gain is applied.
|
||||
bool enable_digital_adaptive = true;
|
||||
// Amount the microphone level is lowered with every clipping event.
|
||||
// Limited to (0, 255].
|
||||
int clipped_level_step = 15;
|
||||
// Proportion of clipped samples required to declare a clipping event.
|
||||
// Limited to (0.f, 1.f).
|
||||
float clipped_ratio_threshold = 0.1f;
|
||||
// Time in frames to wait after a clipping event before checking again.
|
||||
// Limited to values higher than 0.
|
||||
int clipped_wait_frames = 300;
|
||||
|
||||
// Enables clipping prediction functionality.
|
||||
struct ClippingPredictor {
|
||||
bool enabled = false;
|
||||
enum Mode {
|
||||
// Clipping event prediction mode with fixed step estimation.
|
||||
kClippingEventPrediction,
|
||||
// Clipped peak estimation mode with adaptive step estimation.
|
||||
kAdaptiveStepClippingPeakPrediction,
|
||||
// Clipped peak estimation mode with fixed step estimation.
|
||||
kFixedStepClippingPeakPrediction,
|
||||
};
|
||||
Mode mode = kClippingEventPrediction;
|
||||
// Number of frames in the sliding analysis window.
|
||||
int window_length = 5;
|
||||
// Number of frames in the sliding reference window.
|
||||
int reference_window_length = 5;
|
||||
// Reference window delay (unit: number of frames).
|
||||
int reference_window_delay = 5;
|
||||
// Clipping prediction threshold (dBFS).
|
||||
float clipping_threshold = -1.0f;
|
||||
// Crest factor drop threshold (dB).
|
||||
float crest_factor_margin = 3.0f;
|
||||
// If true, the recommended clipped level step is used to modify the
|
||||
// analog gain. Otherwise, the predictor runs without affecting the
|
||||
// analog gain.
|
||||
bool use_predicted_step = true;
|
||||
} clipping_predictor;
|
||||
} analog_gain_controller;
|
||||
} gain_controller1;
|
||||
|
||||
// Parameters for AGC2, an Automatic Gain Control (AGC) sub-module which
|
||||
// replaces the AGC sub-module parametrized by `gain_controller1`.
|
||||
// AGC2 brings the captured audio signal to the desired level by combining
|
||||
// three different controllers (namely, input volume controller, adapative
|
||||
// digital controller and fixed digital controller) and a limiter.
|
||||
// TODO(bugs.webrtc.org:7494): Name `GainController` when AGC1 removed.
|
||||
struct RTC_EXPORT GainController2 {
|
||||
bool operator==(const GainController2& rhs) const;
|
||||
bool operator!=(const GainController2& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
|
||||
// AGC2 must be created if and only if `enabled` is true.
|
||||
bool enabled = false;
|
||||
|
||||
// Parameters for the input volume controller, which adjusts the input
|
||||
// volume applied when the audio is captured (e.g., microphone volume on
|
||||
// a soundcard, input volume on HAL).
|
||||
struct InputVolumeController {
|
||||
bool operator==(const InputVolumeController& rhs) const;
|
||||
bool operator!=(const InputVolumeController& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
bool enabled = false;
|
||||
} input_volume_controller;
|
||||
|
||||
// Parameters for the adaptive digital controller, which adjusts and
|
||||
// applies a digital gain after echo cancellation and after noise
|
||||
// suppression.
|
||||
struct RTC_EXPORT AdaptiveDigital {
|
||||
bool operator==(const AdaptiveDigital& rhs) const;
|
||||
bool operator!=(const AdaptiveDigital& rhs) const {
|
||||
return !(*this == rhs);
|
||||
}
|
||||
bool enabled = false;
|
||||
float headroom_db = 5.0f;
|
||||
float max_gain_db = 50.0f;
|
||||
float initial_gain_db = 15.0f;
|
||||
float max_gain_change_db_per_second = 6.0f;
|
||||
float max_output_noise_level_dbfs = -50.0f;
|
||||
} adaptive_digital;
|
||||
|
||||
// Parameters for the fixed digital controller, which applies a fixed
|
||||
// digital gain after the adaptive digital controller and before the
|
||||
// limiter.
|
||||
struct FixedDigital {
|
||||
// By setting `gain_db` to a value greater than zero, the limiter can be
|
||||
// turned into a compressor that first applies a fixed gain.
|
||||
float gain_db = 0.0f;
|
||||
} fixed_digital;
|
||||
} gain_controller2;
|
||||
|
||||
std::string ToString() const;
|
||||
};
|
||||
|
||||
// Specifies the properties of a setting to be passed to AudioProcessing at
|
||||
// runtime.
|
||||
class RuntimeSetting {
|
||||
public:
|
||||
enum class Type {
|
||||
kNotSpecified,
|
||||
kCapturePreGain,
|
||||
kCaptureCompressionGain,
|
||||
kCaptureFixedPostGain,
|
||||
kPlayoutVolumeChange,
|
||||
kCustomRenderProcessingRuntimeSetting,
|
||||
kPlayoutAudioDeviceChange,
|
||||
kCapturePostGain,
|
||||
kCaptureOutputUsed
|
||||
};
|
||||
|
||||
// Play-out audio device properties.
|
||||
struct PlayoutAudioDeviceInfo {
|
||||
int id; // Identifies the audio device.
|
||||
int max_volume; // Maximum play-out volume.
|
||||
};
|
||||
|
||||
RuntimeSetting() : type_(Type::kNotSpecified), value_(0.0f) {}
|
||||
~RuntimeSetting() = default;
|
||||
|
||||
static RuntimeSetting CreateCapturePreGain(float gain) {
|
||||
return {Type::kCapturePreGain, gain};
|
||||
}
|
||||
|
||||
static RuntimeSetting CreateCapturePostGain(float gain) {
|
||||
return {Type::kCapturePostGain, gain};
|
||||
}
|
||||
|
||||
// Corresponds to Config::GainController1::compression_gain_db, but for
|
||||
// runtime configuration.
|
||||
static RuntimeSetting CreateCompressionGainDb(int gain_db) {
|
||||
RTC_DCHECK_GE(gain_db, 0);
|
||||
RTC_DCHECK_LE(gain_db, 90);
|
||||
return {Type::kCaptureCompressionGain, static_cast<float>(gain_db)};
|
||||
}
|
||||
|
||||
// Corresponds to Config::GainController2::fixed_digital::gain_db, but for
|
||||
// runtime configuration.
|
||||
static RuntimeSetting CreateCaptureFixedPostGain(float gain_db) {
|
||||
RTC_DCHECK_GE(gain_db, 0.0f);
|
||||
RTC_DCHECK_LE(gain_db, 90.0f);
|
||||
return {Type::kCaptureFixedPostGain, gain_db};
|
||||
}
|
||||
|
||||
// Creates a runtime setting to notify play-out (aka render) audio device
|
||||
// changes.
|
||||
static RuntimeSetting CreatePlayoutAudioDeviceChange(
|
||||
PlayoutAudioDeviceInfo audio_device) {
|
||||
return {Type::kPlayoutAudioDeviceChange, audio_device};
|
||||
}
|
||||
|
||||
// Creates a runtime setting to notify play-out (aka render) volume changes.
|
||||
// `volume` is the unnormalized volume, the maximum of which
|
||||
static RuntimeSetting CreatePlayoutVolumeChange(int volume) {
|
||||
return {Type::kPlayoutVolumeChange, volume};
|
||||
}
|
||||
|
||||
static RuntimeSetting CreateCustomRenderSetting(float payload) {
|
||||
return {Type::kCustomRenderProcessingRuntimeSetting, payload};
|
||||
}
|
||||
|
||||
static RuntimeSetting CreateCaptureOutputUsedSetting(
|
||||
bool capture_output_used) {
|
||||
return {Type::kCaptureOutputUsed, capture_output_used};
|
||||
}
|
||||
|
||||
Type type() const { return type_; }
|
||||
// Getters do not return a value but instead modify the argument to protect
|
||||
// from implicit casting.
|
||||
void GetFloat(float* value) const {
|
||||
RTC_DCHECK(value);
|
||||
*value = value_.float_value;
|
||||
}
|
||||
void GetInt(int* value) const {
|
||||
RTC_DCHECK(value);
|
||||
*value = value_.int_value;
|
||||
}
|
||||
void GetBool(bool* value) const {
|
||||
RTC_DCHECK(value);
|
||||
*value = value_.bool_value;
|
||||
}
|
||||
void GetPlayoutAudioDeviceInfo(PlayoutAudioDeviceInfo* value) const {
|
||||
RTC_DCHECK(value);
|
||||
*value = value_.playout_audio_device_info;
|
||||
}
|
||||
|
||||
private:
|
||||
RuntimeSetting(Type id, float value) : type_(id), value_(value) {}
|
||||
RuntimeSetting(Type id, int value) : type_(id), value_(value) {}
|
||||
RuntimeSetting(Type id, PlayoutAudioDeviceInfo value)
|
||||
: type_(id), value_(value) {}
|
||||
Type type_;
|
||||
union U {
|
||||
U() {}
|
||||
U(int value) : int_value(value) {}
|
||||
U(float value) : float_value(value) {}
|
||||
U(PlayoutAudioDeviceInfo value) : playout_audio_device_info(value) {}
|
||||
float float_value;
|
||||
int int_value;
|
||||
bool bool_value;
|
||||
PlayoutAudioDeviceInfo playout_audio_device_info;
|
||||
} value_;
|
||||
};
|
||||
|
||||
~AudioProcessing() override {}
|
||||
|
||||
// Initializes internal states, while retaining all user settings. This
|
||||
// should be called before beginning to process a new audio stream. However,
|
||||
// it is not necessary to call before processing the first stream after
|
||||
// creation.
|
||||
//
|
||||
// It is also not necessary to call if the audio parameters (sample
|
||||
// rate and number of channels) have changed. Passing updated parameters
|
||||
// directly to `ProcessStream()` and `ProcessReverseStream()` is permissible.
|
||||
// If the parameters are known at init-time though, they may be provided.
|
||||
// TODO(webrtc:5298): Change to return void.
|
||||
virtual int Initialize() = 0;
|
||||
|
||||
// The int16 interfaces require:
|
||||
// - only `NativeRate`s be used
|
||||
// - that the input, output and reverse rates must match
|
||||
// - that `processing_config.output_stream()` matches
|
||||
// `processing_config.input_stream()`.
|
||||
//
|
||||
// The float interfaces accept arbitrary rates and support differing input and
|
||||
// output layouts, but the output must have either one channel or the same
|
||||
// number of channels as the input.
|
||||
virtual int Initialize(const ProcessingConfig& processing_config) = 0;
|
||||
|
||||
// TODO(peah): This method is a temporary solution used to take control
|
||||
// over the parameters in the audio processing module and is likely to change.
|
||||
virtual void ApplyConfig(const Config& config) = 0;
|
||||
|
||||
// TODO(ajm): Only intended for internal use. Make private and friend the
|
||||
// necessary classes?
|
||||
virtual int proc_sample_rate_hz() const = 0;
|
||||
virtual int proc_split_sample_rate_hz() const = 0;
|
||||
virtual size_t num_input_channels() const = 0;
|
||||
virtual size_t num_proc_channels() const = 0;
|
||||
virtual size_t num_output_channels() const = 0;
|
||||
virtual size_t num_reverse_channels() const = 0;
|
||||
|
||||
// Set to true when the output of AudioProcessing will be muted or in some
|
||||
// other way not used. Ideally, the captured audio would still be processed,
|
||||
// but some components may change behavior based on this information.
|
||||
// Default false. This method takes a lock. To achieve this in a lock-less
|
||||
// manner the PostRuntimeSetting can instead be used.
|
||||
virtual void set_output_will_be_muted(bool muted) = 0;
|
||||
|
||||
// Enqueues a runtime setting.
|
||||
virtual void SetRuntimeSetting(RuntimeSetting setting) = 0;
|
||||
|
||||
// Enqueues a runtime setting. Returns a bool indicating whether the
|
||||
// enqueueing was successfull.
|
||||
virtual bool PostRuntimeSetting(RuntimeSetting setting) = 0;
|
||||
|
||||
// Accepts and produces a ~10 ms frame of interleaved 16 bit integer audio as
|
||||
// specified in `input_config` and `output_config`. `src` and `dest` may use
|
||||
// the same memory, if desired.
|
||||
virtual int ProcessStream(const int16_t* const src,
|
||||
const StreamConfig& input_config,
|
||||
const StreamConfig& output_config,
|
||||
int16_t* const dest) = 0;
|
||||
|
||||
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
|
||||
// `src` points to a channel buffer, arranged according to `input_stream`. At
|
||||
// output, the channels will be arranged according to `output_stream` in
|
||||
// `dest`.
|
||||
//
|
||||
// The output must have one channel or as many channels as the input. `src`
|
||||
// and `dest` may use the same memory, if desired.
|
||||
virtual int ProcessStream(const float* const* src,
|
||||
const StreamConfig& input_config,
|
||||
const StreamConfig& output_config,
|
||||
float* const* dest) = 0;
|
||||
|
||||
// Accepts and produces a ~10 ms frame of interleaved 16 bit integer audio for
|
||||
// the reverse direction audio stream as specified in `input_config` and
|
||||
// `output_config`. `src` and `dest` may use the same memory, if desired.
|
||||
virtual int ProcessReverseStream(const int16_t* const src,
|
||||
const StreamConfig& input_config,
|
||||
const StreamConfig& output_config,
|
||||
int16_t* const dest) = 0;
|
||||
|
||||
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
|
||||
// `data` points to a channel buffer, arranged according to `reverse_config`.
|
||||
virtual int ProcessReverseStream(const float* const* src,
|
||||
const StreamConfig& input_config,
|
||||
const StreamConfig& output_config,
|
||||
float* const* dest) = 0;
|
||||
|
||||
// Accepts deinterleaved float audio with the range [-1, 1]. Each element
|
||||
// of `data` points to a channel buffer, arranged according to
|
||||
// `reverse_config`.
|
||||
virtual int AnalyzeReverseStream(const float* const* data,
|
||||
const StreamConfig& reverse_config) = 0;
|
||||
|
||||
// Returns the most recently produced ~10 ms of the linear AEC output at a
|
||||
// rate of 16 kHz. If there is more than one capture channel, a mono
|
||||
// representation of the input is returned. Returns true/false to indicate
|
||||
// whether an output returned.
|
||||
virtual bool GetLinearAecOutput(
|
||||
rtc::ArrayView<std::array<float, 160>> linear_output) const = 0;
|
||||
|
||||
// This must be called prior to ProcessStream() if and only if adaptive analog
|
||||
// gain control is enabled, to pass the current analog level from the audio
|
||||
// HAL. Must be within the range [0, 255].
|
||||
virtual void set_stream_analog_level(int level) = 0;
|
||||
|
||||
// When an analog mode is set, this should be called after
|
||||
// `set_stream_analog_level()` and `ProcessStream()` to obtain the recommended
|
||||
// new analog level for the audio HAL. It is the user's responsibility to
|
||||
// apply this level.
|
||||
virtual int recommended_stream_analog_level() const = 0;
|
||||
|
||||
// This must be called if and only if echo processing is enabled.
|
||||
//
|
||||
// Sets the `delay` in ms between ProcessReverseStream() receiving a far-end
|
||||
// frame and ProcessStream() receiving a near-end frame containing the
|
||||
// corresponding echo. On the client-side this can be expressed as
|
||||
// delay = (t_render - t_analyze) + (t_process - t_capture)
|
||||
// where,
|
||||
// - t_analyze is the time a frame is passed to ProcessReverseStream() and
|
||||
// t_render is the time the first sample of the same frame is rendered by
|
||||
// the audio hardware.
|
||||
// - t_capture is the time the first sample of a frame is captured by the
|
||||
// audio hardware and t_process is the time the same frame is passed to
|
||||
// ProcessStream().
|
||||
virtual int set_stream_delay_ms(int delay) = 0;
|
||||
virtual int stream_delay_ms() const = 0;
|
||||
|
||||
// Call to signal that a key press occurred (true) or did not occur (false)
|
||||
// with this chunk of audio.
|
||||
virtual void set_stream_key_pressed(bool key_pressed) = 0;
|
||||
|
||||
// Creates and attaches an webrtc::AecDump for recording debugging
|
||||
// information.
|
||||
// The `worker_queue` may not be null and must outlive the created
|
||||
// AecDump instance. |max_log_size_bytes == -1| means the log size
|
||||
// will be unlimited. `handle` may not be null. The AecDump takes
|
||||
// responsibility for `handle` and closes it in the destructor. A
|
||||
// return value of true indicates that the file has been
|
||||
// sucessfully opened, while a value of false indicates that
|
||||
// opening the file failed.
|
||||
virtual bool CreateAndAttachAecDump(
|
||||
absl::string_view file_name,
|
||||
int64_t max_log_size_bytes,
|
||||
absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
|
||||
virtual bool CreateAndAttachAecDump(
|
||||
absl::Nonnull<FILE*> handle,
|
||||
int64_t max_log_size_bytes,
|
||||
absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
|
||||
|
||||
// TODO(webrtc:5298) Deprecated variant.
|
||||
// Attaches provided webrtc::AecDump for recording debugging
|
||||
// information. Log file and maximum file size logic is supposed to
|
||||
// be handled by implementing instance of AecDump. Calling this
|
||||
// method when another AecDump is attached resets the active AecDump
|
||||
// with a new one. This causes the d-tor of the earlier AecDump to
|
||||
// be called. The d-tor call may block until all pending logging
|
||||
// tasks are completed.
|
||||
virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) = 0;
|
||||
|
||||
// If no AecDump is attached, this has no effect. If an AecDump is
|
||||
// attached, it's destructor is called. The d-tor may block until
|
||||
// all pending logging tasks are completed.
|
||||
virtual void DetachAecDump() = 0;
|
||||
|
||||
// Get audio processing statistics.
|
||||
virtual AudioProcessingStats GetStatistics() = 0;
|
||||
// TODO(webrtc:5298) Deprecated variant. The `has_remote_tracks` argument
|
||||
// should be set if there are active remote tracks (this would usually be true
|
||||
// during a call). If there are no remote tracks some of the stats will not be
|
||||
// set by AudioProcessing, because they only make sense if there is at least
|
||||
// one remote track.
|
||||
virtual AudioProcessingStats GetStatistics(bool has_remote_tracks) = 0;
|
||||
|
||||
// Returns the last applied configuration.
|
||||
virtual AudioProcessing::Config GetConfig() const = 0;
|
||||
|
||||
enum Error {
|
||||
// Fatal errors.
|
||||
kNoError = 0,
|
||||
kUnspecifiedError = -1,
|
||||
kCreationFailedError = -2,
|
||||
kUnsupportedComponentError = -3,
|
||||
kUnsupportedFunctionError = -4,
|
||||
kNullPointerError = -5,
|
||||
kBadParameterError = -6,
|
||||
kBadSampleRateError = -7,
|
||||
kBadDataLengthError = -8,
|
||||
kBadNumberChannelsError = -9,
|
||||
kFileError = -10,
|
||||
kStreamParameterNotSetError = -11,
|
||||
kNotEnabledError = -12,
|
||||
|
||||
// Warnings are non-fatal.
|
||||
// This results when a set_stream_ parameter is out of range. Processing
|
||||
// will continue, but the parameter may have been truncated.
|
||||
kBadStreamParameterWarning = -13
|
||||
};
|
||||
|
||||
// Native rates supported by the integer interfaces.
|
||||
enum NativeRate {
|
||||
kSampleRate8kHz = 8000,
|
||||
kSampleRate16kHz = 16000,
|
||||
kSampleRate32kHz = 32000,
|
||||
kSampleRate48kHz = 48000
|
||||
};
|
||||
|
||||
// TODO(kwiberg): We currently need to support a compiler (Visual C++) that
|
||||
// complains if we don't explicitly state the size of the array here. Remove
|
||||
// the size when that's no longer the case.
|
||||
static constexpr int kNativeSampleRatesHz[4] = {
|
||||
kSampleRate8kHz, kSampleRate16kHz, kSampleRate32kHz, kSampleRate48kHz};
|
||||
static constexpr size_t kNumNativeSampleRates =
|
||||
arraysize(kNativeSampleRatesHz);
|
||||
static constexpr int kMaxNativeSampleRateHz =
|
||||
kNativeSampleRatesHz[kNumNativeSampleRates - 1];
|
||||
|
||||
// APM processes audio in chunks of about 10 ms. See GetFrameSize() for
|
||||
// details.
|
||||
static constexpr int kChunkSizeMs = 10;
|
||||
|
||||
// Returns floor(sample_rate_hz/100): the number of samples per channel used
|
||||
// as input and output to the audio processing module in calls to
|
||||
// ProcessStream, ProcessReverseStream, AnalyzeReverseStream, and
|
||||
// GetLinearAecOutput.
|
||||
//
|
||||
// This is exactly 10 ms for sample rates divisible by 100. For example:
|
||||
// - 48000 Hz (480 samples per channel),
|
||||
// - 44100 Hz (441 samples per channel),
|
||||
// - 16000 Hz (160 samples per channel).
|
||||
//
|
||||
// Sample rates not divisible by 100 are received/produced in frames of
|
||||
// approximately 10 ms. For example:
|
||||
// - 22050 Hz (220 samples per channel, or ~9.98 ms per frame),
|
||||
// - 11025 Hz (110 samples per channel, or ~9.98 ms per frame).
|
||||
// These nondivisible sample rates yield lower audio quality compared to
|
||||
// multiples of 100. Internal resampling to 10 ms frames causes a simulated
|
||||
// clock drift effect which impacts the performance of (for example) echo
|
||||
// cancellation.
|
||||
static int GetFrameSize(int sample_rate_hz) { return sample_rate_hz / 100; }
|
||||
};
|
||||
|
||||
class RTC_EXPORT AudioProcessingBuilder {
|
||||
public:
|
||||
AudioProcessingBuilder();
|
||||
AudioProcessingBuilder(const AudioProcessingBuilder&) = delete;
|
||||
AudioProcessingBuilder& operator=(const AudioProcessingBuilder&) = delete;
|
||||
~AudioProcessingBuilder();
|
||||
|
||||
// Sets the APM configuration.
|
||||
AudioProcessingBuilder& SetConfig(const AudioProcessing::Config& config) {
|
||||
config_ = config;
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the echo controller factory to inject when APM is created.
|
||||
AudioProcessingBuilder& SetEchoControlFactory(
|
||||
std::unique_ptr<EchoControlFactory> echo_control_factory) {
|
||||
echo_control_factory_ = std::move(echo_control_factory);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the capture post-processing sub-module to inject when APM is created.
|
||||
AudioProcessingBuilder& SetCapturePostProcessing(
|
||||
std::unique_ptr<CustomProcessing> capture_post_processing) {
|
||||
capture_post_processing_ = std::move(capture_post_processing);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the render pre-processing sub-module to inject when APM is created.
|
||||
AudioProcessingBuilder& SetRenderPreProcessing(
|
||||
std::unique_ptr<CustomProcessing> render_pre_processing) {
|
||||
render_pre_processing_ = std::move(render_pre_processing);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the echo detector to inject when APM is created.
|
||||
AudioProcessingBuilder& SetEchoDetector(
|
||||
rtc::scoped_refptr<EchoDetector> echo_detector) {
|
||||
echo_detector_ = std::move(echo_detector);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the capture analyzer sub-module to inject when APM is created.
|
||||
AudioProcessingBuilder& SetCaptureAnalyzer(
|
||||
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer) {
|
||||
capture_analyzer_ = std::move(capture_analyzer);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Creates an APM instance with the specified config or the default one if
|
||||
// unspecified. Injects the specified components transferring the ownership
|
||||
// to the newly created APM instance - i.e., except for the config, the
|
||||
// builder is reset to its initial state.
|
||||
rtc::scoped_refptr<AudioProcessing> Create();
|
||||
|
||||
private:
|
||||
AudioProcessing::Config config_;
|
||||
std::unique_ptr<EchoControlFactory> echo_control_factory_;
|
||||
std::unique_ptr<CustomProcessing> capture_post_processing_;
|
||||
std::unique_ptr<CustomProcessing> render_pre_processing_;
|
||||
rtc::scoped_refptr<EchoDetector> echo_detector_;
|
||||
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer_;
|
||||
};
|
||||
|
||||
class StreamConfig {
|
||||
public:
|
||||
// sample_rate_hz: The sampling rate of the stream.
|
||||
// num_channels: The number of audio channels in the stream.
|
||||
StreamConfig(int sample_rate_hz = 0, size_t num_channels = 0)
|
||||
: sample_rate_hz_(sample_rate_hz),
|
||||
num_channels_(num_channels),
|
||||
num_frames_(calculate_frames(sample_rate_hz)) {}
|
||||
|
||||
void set_sample_rate_hz(int value) {
|
||||
sample_rate_hz_ = value;
|
||||
num_frames_ = calculate_frames(value);
|
||||
}
|
||||
void set_num_channels(size_t value) { num_channels_ = value; }
|
||||
|
||||
int sample_rate_hz() const { return sample_rate_hz_; }
|
||||
|
||||
// The number of channels in the stream.
|
||||
size_t num_channels() const { return num_channels_; }
|
||||
|
||||
size_t num_frames() const { return num_frames_; }
|
||||
size_t num_samples() const { return num_channels_ * num_frames_; }
|
||||
|
||||
bool operator==(const StreamConfig& other) const {
|
||||
return sample_rate_hz_ == other.sample_rate_hz_ &&
|
||||
num_channels_ == other.num_channels_;
|
||||
}
|
||||
|
||||
bool operator!=(const StreamConfig& other) const { return !(*this == other); }
|
||||
|
||||
private:
|
||||
static size_t calculate_frames(int sample_rate_hz) {
|
||||
return static_cast<size_t>(AudioProcessing::GetFrameSize(sample_rate_hz));
|
||||
}
|
||||
|
||||
int sample_rate_hz_;
|
||||
size_t num_channels_;
|
||||
size_t num_frames_;
|
||||
};
|
||||
|
||||
class ProcessingConfig {
|
||||
public:
|
||||
enum StreamName {
|
||||
kInputStream,
|
||||
kOutputStream,
|
||||
kReverseInputStream,
|
||||
kReverseOutputStream,
|
||||
kNumStreamNames,
|
||||
};
|
||||
|
||||
const StreamConfig& input_stream() const {
|
||||
return streams[StreamName::kInputStream];
|
||||
}
|
||||
const StreamConfig& output_stream() const {
|
||||
return streams[StreamName::kOutputStream];
|
||||
}
|
||||
const StreamConfig& reverse_input_stream() const {
|
||||
return streams[StreamName::kReverseInputStream];
|
||||
}
|
||||
const StreamConfig& reverse_output_stream() const {
|
||||
return streams[StreamName::kReverseOutputStream];
|
||||
}
|
||||
|
||||
StreamConfig& input_stream() { return streams[StreamName::kInputStream]; }
|
||||
StreamConfig& output_stream() { return streams[StreamName::kOutputStream]; }
|
||||
StreamConfig& reverse_input_stream() {
|
||||
return streams[StreamName::kReverseInputStream];
|
||||
}
|
||||
StreamConfig& reverse_output_stream() {
|
||||
return streams[StreamName::kReverseOutputStream];
|
||||
}
|
||||
|
||||
bool operator==(const ProcessingConfig& other) const {
|
||||
for (int i = 0; i < StreamName::kNumStreamNames; ++i) {
|
||||
if (this->streams[i] != other.streams[i]) {
|
||||
return false;
|
||||
}
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
||||
bool operator!=(const ProcessingConfig& other) const {
|
||||
return !(*this == other);
|
||||
}
|
||||
|
||||
StreamConfig streams[StreamName::kNumStreamNames];
|
||||
};
|
||||
|
||||
// Experimental interface for a custom analysis submodule.
|
||||
class CustomAudioAnalyzer {
|
||||
public:
|
||||
// (Re-) Initializes the submodule.
|
||||
virtual void Initialize(int sample_rate_hz, int num_channels) = 0;
|
||||
// Analyzes the given capture or render signal.
|
||||
virtual void Analyze(const AudioBuffer* audio) = 0;
|
||||
// Returns a string representation of the module state.
|
||||
virtual std::string ToString() const = 0;
|
||||
|
||||
virtual ~CustomAudioAnalyzer() {}
|
||||
};
|
||||
|
||||
// Interface for a custom processing submodule.
|
||||
class CustomProcessing {
|
||||
public:
|
||||
// (Re-)Initializes the submodule.
|
||||
virtual void Initialize(int sample_rate_hz, int num_channels) = 0;
|
||||
// Processes the given capture or render signal.
|
||||
virtual void Process(AudioBuffer* audio) = 0;
|
||||
// Returns a string representation of the module state.
|
||||
virtual std::string ToString() const = 0;
|
||||
// Handles RuntimeSettings. TODO(webrtc:9262): make pure virtual
|
||||
// after updating dependencies.
|
||||
virtual void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting);
|
||||
|
||||
virtual ~CustomProcessing() {}
|
||||
};
|
||||
|
||||
// Interface for an echo detector submodule.
|
||||
class EchoDetector : public RefCountInterface {
|
||||
public:
|
||||
// (Re-)Initializes the submodule.
|
||||
virtual void Initialize(int capture_sample_rate_hz,
|
||||
int num_capture_channels,
|
||||
int render_sample_rate_hz,
|
||||
int num_render_channels) = 0;
|
||||
|
||||
// Analysis (not changing) of the first channel of the render signal.
|
||||
virtual void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) = 0;
|
||||
|
||||
// Analysis (not changing) of the capture signal.
|
||||
virtual void AnalyzeCaptureAudio(
|
||||
rtc::ArrayView<const float> capture_audio) = 0;
|
||||
|
||||
struct Metrics {
|
||||
absl::optional<double> echo_likelihood;
|
||||
absl::optional<double> echo_likelihood_recent_max;
|
||||
};
|
||||
|
||||
// Collect current metrics from the echo detector.
|
||||
virtual Metrics GetMetrics() const = 0;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
// This is a transitional header forwarding to the new version in the api/
|
||||
// folder.
|
||||
#include "api/audio/audio_processing.h"
|
||||
|
||||
#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
|
||||
|
|
|
|||
|
|
@ -11,57 +11,8 @@
|
|||
#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
|
||||
#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
|
||||
|
||||
#include <stdint.h>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "rtc_base/system/rtc_export.h"
|
||||
|
||||
namespace webrtc {
|
||||
// This version of the stats uses Optionals, it will replace the regular
|
||||
// AudioProcessingStatistics struct.
|
||||
struct RTC_EXPORT AudioProcessingStats {
|
||||
AudioProcessingStats();
|
||||
AudioProcessingStats(const AudioProcessingStats& other);
|
||||
~AudioProcessingStats();
|
||||
|
||||
// Deprecated.
|
||||
// TODO(bugs.webrtc.org/11226): Remove.
|
||||
// True if voice is detected in the last capture frame, after processing.
|
||||
// It is conservative in flagging audio as speech, with low likelihood of
|
||||
// incorrectly flagging a frame as voice.
|
||||
// Only reported if voice detection is enabled in AudioProcessing::Config.
|
||||
absl::optional<bool> voice_detected;
|
||||
|
||||
// AEC Statistics.
|
||||
// ERL = 10log_10(P_far / P_echo)
|
||||
absl::optional<double> echo_return_loss;
|
||||
// ERLE = 10log_10(P_echo / P_out)
|
||||
absl::optional<double> echo_return_loss_enhancement;
|
||||
// Fraction of time that the AEC linear filter is divergent, in a 1-second
|
||||
// non-overlapped aggregation window.
|
||||
absl::optional<double> divergent_filter_fraction;
|
||||
|
||||
// The delay metrics consists of the delay median and standard deviation. It
|
||||
// also consists of the fraction of delay estimates that can make the echo
|
||||
// cancellation perform poorly. The values are aggregated until the first
|
||||
// call to `GetStatistics()` and afterwards aggregated and updated every
|
||||
// second. Note that if there are several clients pulling metrics from
|
||||
// `GetStatistics()` during a session the first call from any of them will
|
||||
// change to one second aggregation window for all.
|
||||
absl::optional<int32_t> delay_median_ms;
|
||||
absl::optional<int32_t> delay_standard_deviation_ms;
|
||||
|
||||
// Residual echo detector likelihood.
|
||||
absl::optional<double> residual_echo_likelihood;
|
||||
// Maximum residual echo likelihood from the last time period.
|
||||
absl::optional<double> residual_echo_likelihood_recent_max;
|
||||
|
||||
// The instantaneous delay estimate produced in the AEC. The unit is in
|
||||
// milliseconds and the value is the instantaneous value at the time of the
|
||||
// call to `GetStatistics()`.
|
||||
absl::optional<int32_t> delay_ms;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
// This is a transitional header forwarding to the new version in the api/
|
||||
// folder.
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
|
||||
#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
|
||||
|
|
|
|||
|
|
@ -15,10 +15,10 @@
|
|||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/task_queue/task_queue_base.h"
|
||||
#include "modules/audio_processing/include/aec_dump.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "test/gmock.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -15,11 +15,11 @@
|
|||
#include <vector>
|
||||
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/echo_detector/circular_buffer.h"
|
||||
#include "modules/audio_processing/echo_detector/mean_variance_estimator.h"
|
||||
#include "modules/audio_processing/echo_detector/moving_max.h"
|
||||
#include "modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -14,8 +14,8 @@
|
|||
#include <vector>
|
||||
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/audio_buffer.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace test {
|
||||
|
|
|
|||
|
|
@ -16,7 +16,7 @@
|
|||
#include <utility>
|
||||
#include <vector>
|
||||
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
|
|
@ -19,11 +19,11 @@
|
|||
#include <vector>
|
||||
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio/echo_canceller3_factory.h"
|
||||
#include "api/audio/echo_detector_creator.h"
|
||||
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
|
||||
#include "modules/audio_processing/echo_control_mobile_impl.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
||||
#include "modules/audio_processing/test/echo_canceller3_config_json.h"
|
||||
#include "modules/audio_processing/test/fake_recording_device.h"
|
||||
|
|
|
|||
|
|
@ -18,9 +18,9 @@
|
|||
#include <string>
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "common_audio/channel_buffer.h"
|
||||
#include "common_audio/include/audio_util.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/test/api_call_statistics.h"
|
||||
#include "modules/audio_processing/test/fake_recording_device.h"
|
||||
#include "modules/audio_processing/test/test_utils.h"
|
||||
|
|
|
|||
|
|
@ -21,7 +21,7 @@
|
|||
#include "absl/flags/flag.h"
|
||||
#include "absl/flags/parse.h"
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/test/aec_dump_based_simulator.h"
|
||||
#include "modules/audio_processing/test/audio_processing_simulator.h"
|
||||
#include "modules/audio_processing/test/wav_based_simulator.h"
|
||||
|
|
|
|||
|
|
@ -13,7 +13,7 @@
|
|||
|
||||
#include <memory>
|
||||
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace test {
|
||||
|
|
|
|||
|
|
@ -14,8 +14,8 @@
|
|||
#include <memory>
|
||||
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "common_audio/channel_buffer.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
// Generated at build-time by the protobuf compiler.
|
||||
#include "modules/audio_processing/debug.pb.h"
|
||||
|
|
|
|||
|
|
@ -11,7 +11,7 @@
|
|||
#ifndef MODULES_AUDIO_PROCESSING_TEST_RUNTIME_SETTING_UTIL_H_
|
||||
#define MODULES_AUDIO_PROCESSING_TEST_RUNTIME_SETTING_UTIL_H_
|
||||
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/test/protobuf_utils.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -14,8 +14,8 @@
|
|||
#include <memory>
|
||||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "modules/audio_processing/audio_buffer.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "rtc_base/random.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
|||
|
|
@ -21,9 +21,9 @@
|
|||
#include <vector>
|
||||
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "common_audio/channel_buffer.h"
|
||||
#include "common_audio/wav_file.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
|||
15
pc/BUILD.gn
15
pc/BUILD.gn
|
|
@ -907,6 +907,7 @@ rtc_source_set("rtc_stats_collector") {
|
|||
"../api:rtp_parameters",
|
||||
"../api:scoped_refptr",
|
||||
"../api:sequence_checker",
|
||||
"../api/audio:audio_processing_statistics",
|
||||
"../api/task_queue:task_queue",
|
||||
"../api/units:time_delta",
|
||||
"../api/video:video_rtp_headers",
|
||||
|
|
@ -916,7 +917,6 @@ rtc_source_set("rtc_stats_collector") {
|
|||
"../media:media_channel",
|
||||
"../media:media_channel_impl",
|
||||
"../modules/audio_device",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
"../p2p:connection",
|
||||
"../p2p:connection_info",
|
||||
|
|
@ -1225,11 +1225,11 @@ rtc_source_set("legacy_stats_collector") {
|
|||
"../api:rtp_sender_interface",
|
||||
"../api:scoped_refptr",
|
||||
"../api:sequence_checker",
|
||||
"../api/audio:audio_processing_statistics",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/video:video_rtp_headers",
|
||||
"../call:call_interfaces",
|
||||
"../media:media_channel",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../p2p:connection",
|
||||
"../p2p:connection_info",
|
||||
"../p2p:ice_transport_internal",
|
||||
|
|
@ -2146,6 +2146,7 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"../api:rtc_stats_api",
|
||||
"../api:scoped_refptr",
|
||||
"../api/audio:audio_mixer_api",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/audio_codecs:builtin_audio_decoder_factory",
|
||||
"../api/audio_codecs:builtin_audio_encoder_factory",
|
||||
|
|
@ -2164,7 +2165,6 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
|
||||
"../media:rtc_media_tests_utils",
|
||||
"../modules/audio_device:audio_device_api",
|
||||
"../modules/audio_processing:api",
|
||||
"../p2p:basic_port_allocator",
|
||||
"../p2p:connection",
|
||||
"../p2p:p2p_test_utils",
|
||||
|
|
@ -2368,6 +2368,8 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"../api:scoped_refptr",
|
||||
"../api/adaptation:resource_adaptation_api",
|
||||
"../api/audio:audio_mixer_api",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio:audio_processing_statistics",
|
||||
"../api/crypto:frame_decryptor_interface",
|
||||
"../api/crypto:frame_encryptor_interface",
|
||||
"../api/crypto:options",
|
||||
|
|
@ -2403,7 +2405,6 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"../media:rtc_media_config",
|
||||
"../media:stream_params",
|
||||
"../modules/audio_device:audio_device_api",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
"../p2p:basic_port_allocator",
|
||||
"../p2p:connection",
|
||||
|
|
@ -2501,7 +2502,6 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"../media:rtc_audio_video",
|
||||
"../media:rtc_media_tests_utils",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../p2p:p2p_test_utils",
|
||||
"../p2p:rtc_p2p",
|
||||
"../rtc_base:safe_conversions",
|
||||
|
|
@ -2598,6 +2598,7 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"../api:rtp_transceiver_direction",
|
||||
"../api:scoped_refptr",
|
||||
"../api/audio:audio_mixer_api",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/crypto:frame_decryptor_interface",
|
||||
"../api/crypto:frame_encryptor_interface",
|
||||
"../api/crypto:options",
|
||||
|
|
@ -2620,8 +2621,6 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"../media:rtc_media_tests_utils",
|
||||
"../media:stream_params",
|
||||
"../modules/audio_device:audio_device_api",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/audio_processing:audio_processing_statistics",
|
||||
"../modules/audio_processing:audioproc_test_utils",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
"../p2p:basic_port_allocator",
|
||||
|
|
@ -2750,6 +2749,7 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"../api:scoped_refptr",
|
||||
"../api:sequence_checker",
|
||||
"../api/audio:audio_mixer_api",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/environment",
|
||||
"../api/environment:environment_factory",
|
||||
|
|
@ -2780,7 +2780,6 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"../media:video_broadcaster",
|
||||
"../modules/audio_device",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
"../p2p:connection",
|
||||
"../p2p:fake_port_allocator",
|
||||
|
|
|
|||
|
|
@ -22,6 +22,7 @@
|
|||
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/audio_codecs/audio_encoder.h"
|
||||
#include "api/candidate.h"
|
||||
#include "api/data_channel_interface.h"
|
||||
|
|
@ -34,7 +35,6 @@
|
|||
#include "api/video/video_timing.h"
|
||||
#include "call/call.h"
|
||||
#include "media/base/media_channel.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "p2p/base/ice_transport_internal.h"
|
||||
#include "p2p/base/p2p_constants.h"
|
||||
#include "pc/channel.h"
|
||||
|
|
|
|||
|
|
@ -16,6 +16,7 @@
|
|||
|
||||
#include "absl/algorithm/container.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/audio_codecs/audio_encoder.h"
|
||||
#include "api/candidate.h"
|
||||
#include "api/data_channel_interface.h"
|
||||
|
|
@ -25,7 +26,6 @@
|
|||
#include "api/scoped_refptr.h"
|
||||
#include "call/call.h"
|
||||
#include "media/base/media_channel.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "p2p/base/ice_transport_internal.h"
|
||||
#include "pc/media_stream.h"
|
||||
#include "pc/rtp_receiver.h"
|
||||
|
|
|
|||
|
|
@ -20,6 +20,7 @@
|
|||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/candidate.h"
|
||||
|
|
@ -46,7 +47,6 @@
|
|||
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
|
||||
#include "media/base/stream_params.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/p2p_constants.h"
|
||||
#include "p2p/base/port.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
|
|
|
|||
|
|
@ -20,6 +20,7 @@
|
|||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/create_peerconnection_factory.h"
|
||||
|
|
@ -38,7 +39,6 @@
|
|||
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
|
||||
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/fake_port_allocator.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
#include "p2p/base/transport_description.h"
|
||||
|
|
|
|||
|
|
@ -16,6 +16,7 @@
|
|||
#include <vector>
|
||||
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/create_peerconnection_factory.h"
|
||||
|
|
@ -38,7 +39,6 @@
|
|||
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
|
||||
#include "media/base/fake_frame_source.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/fake_port_allocator.h"
|
||||
#include "p2p/base/port.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
|
|
|
|||
|
|
@ -20,6 +20,7 @@
|
|||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/candidate.h"
|
||||
#include "api/ice_transport_interface.h"
|
||||
#include "api/jsep.h"
|
||||
|
|
@ -28,7 +29,6 @@
|
|||
#include "api/rtc_error.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/fake_port_allocator.h"
|
||||
#include "p2p/base/ice_transport_internal.h"
|
||||
#include "p2p/base/p2p_constants.h"
|
||||
|
|
|
|||
|
|
@ -20,6 +20,7 @@
|
|||
#include "absl/strings/str_replace.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/create_peerconnection_factory.h"
|
||||
|
|
@ -55,7 +56,6 @@
|
|||
#include "media/engine/webrtc_media_engine.h"
|
||||
#include "media/sctp/sctp_transport_internal.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/fake_port_allocator.h"
|
||||
#include "p2p/base/p2p_constants.h"
|
||||
#include "p2p/base/port.h"
|
||||
|
|
|
|||
|
|
@ -15,6 +15,7 @@
|
|||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/audio_options.h"
|
||||
|
|
@ -40,7 +41,6 @@
|
|||
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
|
||||
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
#include "p2p/base/port_interface.h"
|
||||
#include "p2p/base/test_turn_server.h"
|
||||
|
|
|
|||
|
|
@ -18,6 +18,7 @@
|
|||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/create_peerconnection_factory.h"
|
||||
|
|
@ -46,7 +47,6 @@
|
|||
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
|
||||
#include "media/base/stream_params.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
#include "pc/media_session.h"
|
||||
#include "pc/peer_connection_wrapper.h"
|
||||
|
|
|
|||
|
|
@ -25,6 +25,7 @@
|
|||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/create_peerconnection_factory.h"
|
||||
|
|
@ -51,7 +52,6 @@
|
|||
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
|
||||
#include "media/base/codec.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
#include "pc/peer_connection.h"
|
||||
#include "pc/peer_connection_proxy.h"
|
||||
|
|
|
|||
|
|
@ -21,6 +21,7 @@
|
|||
#include "absl/strings/match.h"
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/audio_codecs/opus_audio_decoder_factory.h"
|
||||
|
|
@ -51,7 +52,6 @@
|
|||
#include "media/base/rid_description.h"
|
||||
#include "media/base/stream_params.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "pc/channel_interface.h"
|
||||
#include "pc/peer_connection_wrapper.h"
|
||||
#include "pc/sdp_utils.h"
|
||||
|
|
|
|||
|
|
@ -25,6 +25,7 @@
|
|||
#include "absl/functional/bind_front.h"
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/candidate.h"
|
||||
#include "api/dtls_transport_interface.h"
|
||||
#include "api/media_stream_interface.h"
|
||||
|
|
@ -40,7 +41,6 @@
|
|||
#include "media/base/media_channel.h"
|
||||
#include "media/base/media_channel_impl.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "modules/rtp_rtcp/include/report_block_data.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
||||
#include "p2p/base/connection_info.h"
|
||||
|
|
|
|||
|
|
@ -24,6 +24,7 @@
|
|||
|
||||
#include "absl/strings/str_replace.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/candidate.h"
|
||||
#include "api/dtls_transport_interface.h"
|
||||
#include "api/media_stream_interface.h"
|
||||
|
|
@ -46,7 +47,6 @@
|
|||
#include "common_video/include/quality_limitation_reason.h"
|
||||
#include "media/base/media_channel.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing_statistics.h"
|
||||
#include "modules/rtp_rtcp/include/report_block_data.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
||||
#include "p2p/base/connection_info.h"
|
||||
|
|
|
|||
|
|
@ -14,6 +14,7 @@
|
|||
|
||||
#include "absl/strings/str_replace.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/create_peerconnection_factory.h"
|
||||
|
|
@ -32,7 +33,6 @@
|
|||
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
|
||||
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
#include "pc/peer_connection_wrapper.h"
|
||||
#include "pc/session_description.h"
|
||||
|
|
|
|||
|
|
@ -30,6 +30,7 @@
|
|||
#include "absl/memory/memory.h"
|
||||
#include "absl/strings/string_view.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio_options.h"
|
||||
#include "api/candidate.h"
|
||||
#include "api/crypto/crypto_options.h"
|
||||
|
|
@ -69,7 +70,6 @@
|
|||
#include "media/base/stream_params.h"
|
||||
#include "media/engine/fake_webrtc_video_engine.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
|
||||
#include "p2p/base/fake_ice_transport.h"
|
||||
#include "p2p/base/ice_transport_internal.h"
|
||||
|
|
|
|||
|
|
@ -20,6 +20,7 @@
|
|||
#include "absl/strings/match.h"
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/create_peerconnection_factory.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/media_types.h"
|
||||
|
|
@ -38,7 +39,6 @@
|
|||
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
|
||||
#include "media/engine/simulcast_encoder_adapter.h"
|
||||
#include "modules/audio_device/include/audio_device.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
#include "p2p/base/fake_port_allocator.h"
|
||||
#include "p2p/base/port_allocator.h"
|
||||
#include "pc/test/fake_periodic_video_source.h"
|
||||
|
|
|
|||
|
|
@ -639,8 +639,8 @@ if (rtc_include_tests) {
|
|||
sources = [ "audioproc_f/audioproc_float_main.cc" ]
|
||||
deps = [
|
||||
"../api:audioproc_f_api",
|
||||
"../api/audio:audio_processing",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
]
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -10,8 +10,8 @@
|
|||
|
||||
#include <memory>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/test/audioproc_float.h"
|
||||
#include "modules/audio_processing/include/audio_processing.h"
|
||||
|
||||
int main(int argc, char* argv[]) {
|
||||
return webrtc::test::AudioprocFloat(
|
||||
|
|
|
|||
|
|
@ -1066,6 +1066,7 @@ if (is_ios || is_mac) {
|
|||
"../api:rtp_parameters",
|
||||
"../api:rtp_sender_interface",
|
||||
"../api:scoped_refptr",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/audio_codecs:builtin_audio_decoder_factory",
|
||||
"../api/audio_codecs:builtin_audio_encoder_factory",
|
||||
|
|
@ -1082,7 +1083,6 @@ if (is_ios || is_mac) {
|
|||
"../media:rtc_media_base",
|
||||
"../modules/audio_device:audio_device_api",
|
||||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/video_coding:video_codec_interface",
|
||||
"../pc:peer_connection_factory",
|
||||
"../pc:webrtc_sdp",
|
||||
|
|
@ -1169,6 +1169,7 @@ if (is_ios || is_mac) {
|
|||
":videosource_objc",
|
||||
":videotoolbox_objc",
|
||||
"../api:scoped_refptr",
|
||||
"../api/audio:audio_processing",
|
||||
"../api/audio_codecs:builtin_audio_decoder_factory",
|
||||
"../api/audio_codecs:builtin_audio_encoder_factory",
|
||||
"../api/environment:environment_factory",
|
||||
|
|
@ -1180,7 +1181,6 @@ if (is_ios || is_mac) {
|
|||
"../media:rtc_media_base",
|
||||
"../media:rtc_media_tests_utils",
|
||||
"../modules/audio_device:audio_device_api",
|
||||
"../modules/audio_processing:api",
|
||||
"../modules/video_coding:video_codec_interface",
|
||||
"../rtc_base:gunit_helpers",
|
||||
"../rtc_base:macromagic",
|
||||
|
|
|
|||
|
|
@ -573,8 +573,8 @@ if (current_os == "linux" || is_android) {
|
|||
"../../api:field_trials_view",
|
||||
"../../api:scoped_refptr",
|
||||
"../../api:sequence_checker",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../api/task_queue:pending_task_safety_flag",
|
||||
"../../modules/audio_processing:api",
|
||||
"../../rtc_base:checks",
|
||||
"../../rtc_base:ip_address",
|
||||
"../../rtc_base:logging",
|
||||
|
|
@ -598,8 +598,8 @@ if (current_os == "linux" || is_android) {
|
|||
|
||||
deps = [
|
||||
":base_jni",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../modules/audio_processing",
|
||||
"../../modules/audio_processing:api",
|
||||
]
|
||||
}
|
||||
|
||||
|
|
@ -782,6 +782,7 @@ if (current_os == "linux" || is_android) {
|
|||
"../../api:rtp_parameters",
|
||||
"../../api:rtp_sender_interface",
|
||||
"../../api:turn_customizer",
|
||||
"../../api/audio:audio_processing",
|
||||
"../../api/crypto:options",
|
||||
"../../api/rtc_event_log:rtc_event_log_factory",
|
||||
"../../api/task_queue:default_task_queue_factory",
|
||||
|
|
@ -789,7 +790,6 @@ if (current_os == "linux" || is_android) {
|
|||
"../../call:call_interfaces",
|
||||
"../../media:rtc_media_base",
|
||||
"../../modules/audio_device",
|
||||
"../../modules/audio_processing:api",
|
||||
"../../modules/utility",
|
||||
"../../pc:media_stream_observer",
|
||||
"../../pc:peer_connection_factory",
|
||||
|
|
@ -1659,7 +1659,6 @@ if (is_android) {
|
|||
"../../media:rtc_internal_video_codecs",
|
||||
"../../modules/audio_device",
|
||||
"../../modules/audio_device:mock_audio_device",
|
||||
"../../modules/audio_processing:api",
|
||||
"../../modules/utility",
|
||||
"../../pc:libjingle_peerconnection",
|
||||
"../../rtc_base:checks",
|
||||
|
|
|
|||
Some files were not shown because too many files have changed in this diff Show more
Loading…
Reference in a new issue