A simple copy of the old audio mixer to a new directory.
I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer. This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer. NOTRY=True Review-Url: https://codereview.webrtc.org/2104363003 Cr-Commit-Position: refs/heads/master@{#13378}
This commit is contained in:
aleloi
72
webrtc/modules/audio_mixer/BUILD.gn
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72
webrtc/modules/audio_mixer/BUILD.gn
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@ -0,0 +1,72 @@
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|
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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|
# that can be found in the LICENSE file in the root of the source
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|
# tree. An additional intellectual property rights grant can be found
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||||||
|
# in the file PATENTS. All contributing project authors may
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|
# be found in the AUTHORS file in the root of the source tree.
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|
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config("audio_conference_mixer_config") {
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visibility = [ ":*" ] # Only targets in this file can depend on this.
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include_dirs = [
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"include",
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"../../modules/include",
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]
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}
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source_set("audio_mixer") {
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sources = [
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"audio_mixer.cc",
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"audio_mixer.h",
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]
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deps = [
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":audio_conference_mixer",
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"../../voice_engine:voice_engine",
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]
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if (is_win) {
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defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]
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cflags = [
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# TODO(kjellander): Bug 261: fix this warning.
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"/wd4373", # virtual function override.
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]
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}
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configs += [ "../..:common_config" ]
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public_configs = [ "../..:common_inherited_config" ]
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|
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|
if (is_clang) {
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|
# Suppress warnings from Chrome's Clang plugins.
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|
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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|
configs -= [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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source_set("audio_conference_mixer") {
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sources = [
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"include/audio_mixer_defines.h",
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"include/new_audio_conference_mixer.h",
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"source/new_audio_conference_mixer_impl.cc",
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"source/new_audio_conference_mixer_impl.h",
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]
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configs += [ "../..:common_config" ]
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public_configs = [
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"../..:common_inherited_config",
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":audio_conference_mixer_config",
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]
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|
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|
if (is_clang) {
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# Suppress warnings from Chrome's Clang plugins.
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|
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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|
configs -= [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"../../modules/audio_processing",
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"../../modules/utility",
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"../../system_wrappers",
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]
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}
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15
webrtc/modules/audio_mixer/DEPS
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15
webrtc/modules/audio_mixer/DEPS
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|
include_rules = [
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"+webrtc/base",
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"+webrtc/call",
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"+webrtc/common_audio",
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"+webrtc/modules/audio_coding",
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"+webrtc/modules/audio_conference_mixer",
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"+webrtc/modules/audio_device",
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"+webrtc/modules/audio_processing",
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"+webrtc/modules/media_file",
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"+webrtc/modules/pacing",
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"+webrtc/modules/rtp_rtcp",
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"+webrtc/modules/utility",
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|
"+webrtc/system_wrappers",
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"+webrtc/voice_engine",
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|
]
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451
webrtc/modules/audio_mixer/audio_mixer.cc
Normal file
451
webrtc/modules/audio_mixer/audio_mixer.cc
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|
/*
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|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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|
*
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|
* Use of this source code is governed by a BSD-style license
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|
* that can be found in the LICENSE file in the root of the source
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|
* tree. An additional intellectual property rights grant can be found
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|
* in the file PATENTS. All contributing project authors may
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|
* be found in the AUTHORS file in the root of the source tree.
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|
*/
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|
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#include "webrtc/modules/audio_mixer/audio_mixer.h"
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#include "webrtc/base/format_macros.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/utility/include/audio_frame_operations.h"
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#include "webrtc/system_wrappers/include/file_wrapper.h"
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#include "webrtc/system_wrappers/include/trace.h"
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#include "webrtc/voice_engine/include/voe_external_media.h"
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#include "webrtc/voice_engine/statistics.h"
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#include "webrtc/voice_engine/utility.h"
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namespace webrtc {
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namespace voe {
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void AudioMixer::NewMixedAudio(int32_t id,
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const AudioFrame& generalAudioFrame,
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const AudioFrame** uniqueAudioFrames,
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uint32_t size) {
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::NewMixedAudio(id=%d, size=%u)", id, size);
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|
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|
_audioFrame.CopyFrom(generalAudioFrame);
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_audioFrame.id_ = id;
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|
}
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|
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void AudioMixer::PlayNotification(int32_t id, uint32_t durationMs) {
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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|
"AudioMixer::PlayNotification(id=%d, durationMs=%d)", id,
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|
durationMs);
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// Not implement yet
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|
}
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void AudioMixer::RecordNotification(int32_t id, uint32_t durationMs) {
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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|
"AudioMixer::RecordNotification(id=%d, durationMs=%d)", id,
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|
durationMs);
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// Not implement yet
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|
}
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void AudioMixer::PlayFileEnded(int32_t id) {
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|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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|
"AudioMixer::PlayFileEnded(id=%d)", id);
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|
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|
// not needed
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|
}
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void AudioMixer::RecordFileEnded(int32_t id) {
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|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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|
"AudioMixer::RecordFileEnded(id=%d)", id);
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|
assert(id == _instanceId);
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rtc::CritScope cs(&_fileCritSect);
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_outputFileRecording = false;
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WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
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|
"AudioMixer::RecordFileEnded() =>"
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|
"output file recorder module is shutdown");
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|
}
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|
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|
int32_t AudioMixer::Create(AudioMixer*& mixer, uint32_t instanceId) {
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|
WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
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"AudioMixer::Create(instanceId=%d)", instanceId);
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mixer = new AudioMixer(instanceId);
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|
if (mixer == NULL) {
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WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
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|
"AudioMixer::Create() unable to allocate memory for"
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|
"mixer");
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|
return -1;
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|
}
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|
return 0;
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|
}
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|
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|
AudioMixer::AudioMixer(uint32_t instanceId)
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|
: _mixerModule(*NewAudioConferenceMixer::Create(instanceId)),
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|
_audioLevel(),
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|
_instanceId(instanceId),
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|
_externalMediaCallbackPtr(NULL),
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|
_externalMedia(false),
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|
_panLeft(1.0f),
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|
_panRight(1.0f),
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|
_mixingFrequencyHz(8000),
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|
_outputFileRecorderPtr(NULL),
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|
_outputFileRecording(false) {
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|
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
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|
"AudioMixer::AudioMixer() - ctor");
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|
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|
if (_mixerModule.RegisterMixedStreamCallback(this) == -1) {
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|
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
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|
"AudioMixer::AudioMixer() failed to register mixer"
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|
"callbacks");
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|
}
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|
}
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|
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|
void AudioMixer::Destroy(AudioMixer*& mixer) {
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|
if (mixer) {
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|
delete mixer;
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|
mixer = NULL;
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|
}
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|
}
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|
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|
AudioMixer::~AudioMixer() {
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|
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
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|
"AudioMixer::~AudioMixer() - dtor");
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|
if (_externalMedia) {
|
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|
DeRegisterExternalMediaProcessing();
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|
}
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|
{
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|
rtc::CritScope cs(&_fileCritSect);
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|
if (_outputFileRecorderPtr) {
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|
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
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|
_outputFileRecorderPtr->StopRecording();
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|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
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|
_outputFileRecorderPtr = NULL;
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|
}
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|
}
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|
_mixerModule.UnRegisterMixedStreamCallback();
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|
delete &_mixerModule;
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|
}
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|
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|
int32_t AudioMixer::SetEngineInformation(voe::Statistics& engineStatistics) {
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|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
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|
"AudioMixer::SetEngineInformation()");
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|
_engineStatisticsPtr = &engineStatistics;
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|
return 0;
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|
}
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|
|
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|
int32_t AudioMixer::SetAudioProcessingModule(
|
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|
AudioProcessing* audioProcessingModule) {
|
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|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioMixer::SetAudioProcessingModule("
|
||||||
|
"audioProcessingModule=0x%x)",
|
||||||
|
audioProcessingModule);
|
||||||
|
_audioProcessingModulePtr = audioProcessingModule;
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::RegisterExternalMediaProcessing(
|
||||||
|
VoEMediaProcess& proccess_object) {
|
||||||
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioMixer::RegisterExternalMediaProcessing()");
|
||||||
|
|
||||||
|
rtc::CritScope cs(&_callbackCritSect);
|
||||||
|
_externalMediaCallbackPtr = &proccess_object;
|
||||||
|
_externalMedia = true;
|
||||||
|
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::DeRegisterExternalMediaProcessing() {
|
||||||
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioMixer::DeRegisterExternalMediaProcessing()");
|
||||||
|
|
||||||
|
rtc::CritScope cs(&_callbackCritSect);
|
||||||
|
_externalMedia = false;
|
||||||
|
_externalMediaCallbackPtr = NULL;
|
||||||
|
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t AudioMixer::SetMixabilityStatus(MixerAudioSource& participant,
|
||||||
|
bool mixable) {
|
||||||
|
return _mixerModule.SetMixabilityStatus(&participant, mixable);
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t AudioMixer::SetAnonymousMixabilityStatus(MixerAudioSource& participant,
|
||||||
|
bool mixable) {
|
||||||
|
return _mixerModule.SetAnonymousMixabilityStatus(&participant, mixable);
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t AudioMixer::MixActiveChannels() {
|
||||||
|
_mixerModule.Process();
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::GetSpeechOutputLevel(uint32_t& level) {
|
||||||
|
int8_t currentLevel = _audioLevel.Level();
|
||||||
|
level = static_cast<uint32_t>(currentLevel);
|
||||||
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"GetSpeechOutputLevel() => level=%u", level);
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::GetSpeechOutputLevelFullRange(uint32_t& level) {
|
||||||
|
int16_t currentLevel = _audioLevel.LevelFullRange();
|
||||||
|
level = static_cast<uint32_t>(currentLevel);
|
||||||
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"GetSpeechOutputLevelFullRange() => level=%u", level);
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::SetOutputVolumePan(float left, float right) {
|
||||||
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioMixer::SetOutputVolumePan()");
|
||||||
|
_panLeft = left;
|
||||||
|
_panRight = right;
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::GetOutputVolumePan(float& left, float& right) {
|
||||||
|
left = _panLeft;
|
||||||
|
right = _panRight;
|
||||||
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"GetOutputVolumePan() => left=%2.1f, right=%2.1f", left, right);
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::StartRecordingPlayout(const char* fileName,
|
||||||
|
const CodecInst* codecInst) {
|
||||||
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioMixer::StartRecordingPlayout(fileName=%s)", fileName);
|
||||||
|
|
||||||
|
if (_outputFileRecording) {
|
||||||
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"StartRecordingPlayout() is already recording");
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
FileFormats format;
|
||||||
|
const uint32_t notificationTime(0);
|
||||||
|
CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000};
|
||||||
|
|
||||||
|
if ((codecInst != NULL) &&
|
||||||
|
((codecInst->channels < 1) || (codecInst->channels > 2))) {
|
||||||
|
_engineStatisticsPtr->SetLastError(
|
||||||
|
VE_BAD_ARGUMENT, kTraceError,
|
||||||
|
"StartRecordingPlayout() invalid compression");
|
||||||
|
return (-1);
|
||||||
|
}
|
||||||
|
if (codecInst == NULL) {
|
||||||
|
format = kFileFormatPcm16kHzFile;
|
||||||
|
codecInst = &dummyCodec;
|
||||||
|
} else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) ||
|
||||||
|
(STR_CASE_CMP(codecInst->plname, "PCMU") == 0) ||
|
||||||
|
(STR_CASE_CMP(codecInst->plname, "PCMA") == 0)) {
|
||||||
|
format = kFileFormatWavFile;
|
||||||
|
} else {
|
||||||
|
format = kFileFormatCompressedFile;
|
||||||
|
}
|
||||||
|
|
||||||
|
rtc::CritScope cs(&_fileCritSect);
|
||||||
|
|
||||||
|
// Destroy the old instance
|
||||||
|
if (_outputFileRecorderPtr) {
|
||||||
|
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
|
||||||
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
||||||
|
_outputFileRecorderPtr = NULL;
|
||||||
|
}
|
||||||
|
|
||||||
|
_outputFileRecorderPtr =
|
||||||
|
FileRecorder::CreateFileRecorder(_instanceId, (const FileFormats)format);
|
||||||
|
if (_outputFileRecorderPtr == NULL) {
|
||||||
|
_engineStatisticsPtr->SetLastError(
|
||||||
|
VE_INVALID_ARGUMENT, kTraceError,
|
||||||
|
"StartRecordingPlayout() fileRecorder format isnot correct");
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
|
||||||
|
if (_outputFileRecorderPtr->StartRecordingAudioFile(
|
||||||
|
fileName, (const CodecInst&)*codecInst, notificationTime) != 0) {
|
||||||
|
_engineStatisticsPtr->SetLastError(
|
||||||
|
VE_BAD_FILE, kTraceError,
|
||||||
|
"StartRecordingAudioFile() failed to start file recording");
|
||||||
|
_outputFileRecorderPtr->StopRecording();
|
||||||
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
||||||
|
_outputFileRecorderPtr = NULL;
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
_outputFileRecorderPtr->RegisterModuleFileCallback(this);
|
||||||
|
_outputFileRecording = true;
|
||||||
|
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::StartRecordingPlayout(OutStream* stream,
|
||||||
|
const CodecInst* codecInst) {
|
||||||
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioMixer::StartRecordingPlayout()");
|
||||||
|
|
||||||
|
if (_outputFileRecording) {
|
||||||
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"StartRecordingPlayout() is already recording");
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
FileFormats format;
|
||||||
|
const uint32_t notificationTime(0);
|
||||||
|
CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000};
|
||||||
|
|
||||||
|
if (codecInst != NULL && codecInst->channels != 1) {
|
||||||
|
_engineStatisticsPtr->SetLastError(
|
||||||
|
VE_BAD_ARGUMENT, kTraceError,
|
||||||
|
"StartRecordingPlayout() invalid compression");
|
||||||
|
return (-1);
|
||||||
|
}
|
||||||
|
if (codecInst == NULL) {
|
||||||
|
format = kFileFormatPcm16kHzFile;
|
||||||
|
codecInst = &dummyCodec;
|
||||||
|
} else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) ||
|
||||||
|
(STR_CASE_CMP(codecInst->plname, "PCMU") == 0) ||
|
||||||
|
(STR_CASE_CMP(codecInst->plname, "PCMA") == 0)) {
|
||||||
|
format = kFileFormatWavFile;
|
||||||
|
} else {
|
||||||
|
format = kFileFormatCompressedFile;
|
||||||
|
}
|
||||||
|
|
||||||
|
rtc::CritScope cs(&_fileCritSect);
|
||||||
|
|
||||||
|
// Destroy the old instance
|
||||||
|
if (_outputFileRecorderPtr) {
|
||||||
|
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
|
||||||
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
||||||
|
_outputFileRecorderPtr = NULL;
|
||||||
|
}
|
||||||
|
|
||||||
|
_outputFileRecorderPtr =
|
||||||
|
FileRecorder::CreateFileRecorder(_instanceId, (const FileFormats)format);
|
||||||
|
if (_outputFileRecorderPtr == NULL) {
|
||||||
|
_engineStatisticsPtr->SetLastError(
|
||||||
|
VE_INVALID_ARGUMENT, kTraceError,
|
||||||
|
"StartRecordingPlayout() fileRecorder format isnot correct");
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
|
||||||
|
if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, *codecInst,
|
||||||
|
notificationTime) != 0) {
|
||||||
|
_engineStatisticsPtr->SetLastError(
|
||||||
|
VE_BAD_FILE, kTraceError,
|
||||||
|
"StartRecordingAudioFile() failed to start file recording");
|
||||||
|
_outputFileRecorderPtr->StopRecording();
|
||||||
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
||||||
|
_outputFileRecorderPtr = NULL;
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
|
||||||
|
_outputFileRecorderPtr->RegisterModuleFileCallback(this);
|
||||||
|
_outputFileRecording = true;
|
||||||
|
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::StopRecordingPlayout() {
|
||||||
|
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioMixer::StopRecordingPlayout()");
|
||||||
|
|
||||||
|
if (!_outputFileRecording) {
|
||||||
|
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"StopRecordingPlayout() file isnot recording");
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
|
||||||
|
rtc::CritScope cs(&_fileCritSect);
|
||||||
|
|
||||||
|
if (_outputFileRecorderPtr->StopRecording() != 0) {
|
||||||
|
_engineStatisticsPtr->SetLastError(
|
||||||
|
VE_STOP_RECORDING_FAILED, kTraceError,
|
||||||
|
"StopRecording(), could not stop recording");
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
|
||||||
|
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
|
||||||
|
_outputFileRecorderPtr = NULL;
|
||||||
|
_outputFileRecording = false;
|
||||||
|
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int AudioMixer::GetMixedAudio(int sample_rate_hz,
|
||||||
|
size_t num_channels,
|
||||||
|
AudioFrame* frame) {
|
||||||
|
WEBRTC_TRACE(
|
||||||
|
kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%" PRIuS ")",
|
||||||
|
sample_rate_hz, num_channels);
|
||||||
|
|
||||||
|
// --- Record playout if enabled
|
||||||
|
{
|
||||||
|
rtc::CritScope cs(&_fileCritSect);
|
||||||
|
if (_outputFileRecording && _outputFileRecorderPtr)
|
||||||
|
_outputFileRecorderPtr->RecordAudioToFile(_audioFrame);
|
||||||
|
}
|
||||||
|
|
||||||
|
frame->num_channels_ = num_channels;
|
||||||
|
frame->sample_rate_hz_ = sample_rate_hz;
|
||||||
|
// TODO(andrew): Ideally the downmixing would occur much earlier, in
|
||||||
|
// AudioCodingModule.
|
||||||
|
RemixAndResample(_audioFrame, &resampler_, frame);
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t AudioMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm) {
|
||||||
|
if (_audioFrame.sample_rate_hz_ != _mixingFrequencyHz) {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioMixer::DoOperationsOnCombinedSignal() => "
|
||||||
|
"mixing frequency = %d",
|
||||||
|
_audioFrame.sample_rate_hz_);
|
||||||
|
_mixingFrequencyHz = _audioFrame.sample_rate_hz_;
|
||||||
|
}
|
||||||
|
|
||||||
|
// Scale left and/or right channel(s) if balance is active
|
||||||
|
if (_panLeft != 1.0 || _panRight != 1.0) {
|
||||||
|
if (_audioFrame.num_channels_ == 1) {
|
||||||
|
AudioFrameOperations::MonoToStereo(&_audioFrame);
|
||||||
|
} else {
|
||||||
|
// Pure stereo mode (we are receiving a stereo signal).
|
||||||
|
}
|
||||||
|
|
||||||
|
assert(_audioFrame.num_channels_ == 2);
|
||||||
|
AudioFrameOperations::Scale(_panLeft, _panRight, _audioFrame);
|
||||||
|
}
|
||||||
|
|
||||||
|
// --- Far-end Voice Quality Enhancement (AudioProcessing Module)
|
||||||
|
if (feed_data_to_apm) {
|
||||||
|
if (_audioProcessingModulePtr->ProcessReverseStream(&_audioFrame) != 0) {
|
||||||
|
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
|
||||||
|
"AudioProcessingModule::ProcessReverseStream() => error");
|
||||||
|
RTC_DCHECK(false);
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
// --- External media processing
|
||||||
|
{
|
||||||
|
rtc::CritScope cs(&_callbackCritSect);
|
||||||
|
if (_externalMedia) {
|
||||||
|
const bool is_stereo = (_audioFrame.num_channels_ == 2);
|
||||||
|
if (_externalMediaCallbackPtr) {
|
||||||
|
_externalMediaCallbackPtr->Process(
|
||||||
|
-1, kPlaybackAllChannelsMixed,
|
||||||
|
reinterpret_cast<int16_t*>(_audioFrame.data_),
|
||||||
|
_audioFrame.samples_per_channel_, _audioFrame.sample_rate_hz_,
|
||||||
|
is_stereo);
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
// --- Measure audio level (0-9) for the combined signal
|
||||||
|
_audioLevel.ComputeLevel(_audioFrame);
|
||||||
|
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
} // namespace voe
|
||||||
|
} // namespace webrtc
|
||||||
40
webrtc/modules/audio_mixer/audio_mixer.gypi
Normal file
40
webrtc/modules/audio_mixer/audio_mixer.gypi
Normal file
@ -0,0 +1,40 @@
|
|||||||
|
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
|
||||||
|
#
|
||||||
|
# Use of this source code is governed by a BSD-style license
|
||||||
|
# that can be found in the LICENSE file in the root of the source
|
||||||
|
# tree. An additional intellectual property rights grant can be found
|
||||||
|
# in the file PATENTS. All contributing project authors may
|
||||||
|
# be found in the AUTHORS file in the root of the source tree.
|
||||||
|
|
||||||
|
{
|
||||||
|
'targets': [
|
||||||
|
{
|
||||||
|
'target_name': 'new_audio_conference_mixer',
|
||||||
|
'type': 'static_library',
|
||||||
|
'dependencies': [
|
||||||
|
'audio_processing',
|
||||||
|
'webrtc_utility',
|
||||||
|
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
|
||||||
|
],
|
||||||
|
'sources': [
|
||||||
|
'include/new_audio_conference_mixer.h',
|
||||||
|
'include/audio_mixer_defines.h',
|
||||||
|
'source/new_audio_conference_mixer_impl.cc',
|
||||||
|
'source/new_audio_conference_mixer_impl.h',
|
||||||
|
],
|
||||||
|
},
|
||||||
|
{
|
||||||
|
'target_name': 'audio_mixer',
|
||||||
|
'type': 'static_library',
|
||||||
|
'dependencies': [
|
||||||
|
'new_audio_conference_mixer',
|
||||||
|
'webrtc_utility',
|
||||||
|
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
|
||||||
|
],
|
||||||
|
'sources': [
|
||||||
|
'audio_mixer.h',
|
||||||
|
'audio_mixer.cc',
|
||||||
|
],
|
||||||
|
},
|
||||||
|
], # targets
|
||||||
|
}
|
||||||
127
webrtc/modules/audio_mixer/audio_mixer.h
Normal file
127
webrtc/modules/audio_mixer/audio_mixer.h
Normal file
@ -0,0 +1,127 @@
|
|||||||
|
/*
|
||||||
|
* 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 WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_H_
|
||||||
|
#define WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_H_
|
||||||
|
|
||||||
|
#include "webrtc/base/criticalsection.h"
|
||||||
|
#include "webrtc/common_audio/resampler/include/push_resampler.h"
|
||||||
|
#include "webrtc/common_types.h"
|
||||||
|
#include "webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h"
|
||||||
|
#include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
|
||||||
|
#include "webrtc/modules/utility/include/file_recorder.h"
|
||||||
|
#include "webrtc/voice_engine/level_indicator.h"
|
||||||
|
#include "webrtc/voice_engine/voice_engine_defines.h"
|
||||||
|
|
||||||
|
namespace webrtc {
|
||||||
|
|
||||||
|
class AudioProcessing;
|
||||||
|
class FileWrapper;
|
||||||
|
class VoEMediaProcess;
|
||||||
|
|
||||||
|
namespace voe {
|
||||||
|
class Statistics;
|
||||||
|
|
||||||
|
// Note: this class is in the process of being rewritten and merged
|
||||||
|
// with AudioConferenceMixer. Expect inheritance chains to be changed,
|
||||||
|
// member functions removed or renamed.
|
||||||
|
class AudioMixer : public OldAudioMixerOutputReceiver, public FileCallback {
|
||||||
|
public:
|
||||||
|
static int32_t Create(AudioMixer*& mixer, uint32_t instanceId); // NOLINT
|
||||||
|
|
||||||
|
static void Destroy(AudioMixer*& mixer); // NOLINT
|
||||||
|
|
||||||
|
int32_t SetEngineInformation(Statistics& engineStatistics); // NOLINT
|
||||||
|
|
||||||
|
int32_t SetAudioProcessingModule(AudioProcessing* audioProcessingModule);
|
||||||
|
|
||||||
|
// VoEExternalMedia
|
||||||
|
int RegisterExternalMediaProcessing(VoEMediaProcess& // NOLINT
|
||||||
|
proccess_object);
|
||||||
|
|
||||||
|
int DeRegisterExternalMediaProcessing();
|
||||||
|
|
||||||
|
int32_t MixActiveChannels();
|
||||||
|
|
||||||
|
int32_t DoOperationsOnCombinedSignal(bool feed_data_to_apm);
|
||||||
|
|
||||||
|
int32_t SetMixabilityStatus(MixerAudioSource& participant, // NOLINT
|
||||||
|
bool mixable);
|
||||||
|
|
||||||
|
int32_t SetAnonymousMixabilityStatus(MixerAudioSource& participant, // NOLINT
|
||||||
|
bool mixable);
|
||||||
|
|
||||||
|
int GetMixedAudio(int sample_rate_hz,
|
||||||
|
size_t num_channels,
|
||||||
|
AudioFrame* audioFrame);
|
||||||
|
|
||||||
|
// VoEVolumeControl
|
||||||
|
int GetSpeechOutputLevel(uint32_t& level); // NOLINT
|
||||||
|
|
||||||
|
int GetSpeechOutputLevelFullRange(uint32_t& level); // NOLINT
|
||||||
|
|
||||||
|
int SetOutputVolumePan(float left, float right);
|
||||||
|
|
||||||
|
int GetOutputVolumePan(float& left, float& right); // NOLINT
|
||||||
|
|
||||||
|
// VoEFile
|
||||||
|
int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst);
|
||||||
|
|
||||||
|
int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
|
||||||
|
int StopRecordingPlayout();
|
||||||
|
|
||||||
|
virtual ~AudioMixer();
|
||||||
|
|
||||||
|
// from AudioMixerOutputReceiver
|
||||||
|
virtual void NewMixedAudio(int32_t id,
|
||||||
|
const AudioFrame& generalAudioFrame,
|
||||||
|
const AudioFrame** uniqueAudioFrames,
|
||||||
|
uint32_t size);
|
||||||
|
|
||||||
|
// For file recording
|
||||||
|
void PlayNotification(int32_t id, uint32_t durationMs);
|
||||||
|
|
||||||
|
void RecordNotification(int32_t id, uint32_t durationMs);
|
||||||
|
|
||||||
|
void PlayFileEnded(int32_t id);
|
||||||
|
void RecordFileEnded(int32_t id);
|
||||||
|
|
||||||
|
private:
|
||||||
|
explicit AudioMixer(uint32_t instanceId);
|
||||||
|
|
||||||
|
// uses
|
||||||
|
Statistics* _engineStatisticsPtr;
|
||||||
|
AudioProcessing* _audioProcessingModulePtr;
|
||||||
|
|
||||||
|
rtc::CriticalSection _callbackCritSect;
|
||||||
|
// protect the _outputFileRecorderPtr and _outputFileRecording
|
||||||
|
rtc::CriticalSection _fileCritSect;
|
||||||
|
NewAudioConferenceMixer& _mixerModule;
|
||||||
|
AudioFrame _audioFrame;
|
||||||
|
// Converts mixed audio to the audio device output rate.
|
||||||
|
PushResampler<int16_t> resampler_;
|
||||||
|
// Converts mixed audio to the audio processing rate.
|
||||||
|
PushResampler<int16_t> audioproc_resampler_;
|
||||||
|
AudioLevel _audioLevel; // measures audio level for the combined signal
|
||||||
|
int _instanceId;
|
||||||
|
VoEMediaProcess* _externalMediaCallbackPtr;
|
||||||
|
bool _externalMedia;
|
||||||
|
float _panLeft;
|
||||||
|
float _panRight;
|
||||||
|
int _mixingFrequencyHz;
|
||||||
|
FileRecorder* _outputFileRecorderPtr;
|
||||||
|
bool _outputFileRecording;
|
||||||
|
};
|
||||||
|
|
||||||
|
} // namespace voe
|
||||||
|
|
||||||
|
} // namespace webrtc
|
||||||
|
|
||||||
|
#endif // WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_H_
|
||||||
84
webrtc/modules/audio_mixer/include/audio_mixer_defines.h
Normal file
84
webrtc/modules/audio_mixer/include/audio_mixer_defines.h
Normal file
@ -0,0 +1,84 @@
|
|||||||
|
/*
|
||||||
|
* Copyright (c) 2011 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 WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_
|
||||||
|
#define WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_
|
||||||
|
|
||||||
|
#include "webrtc/base/checks.h"
|
||||||
|
#include "webrtc/modules/include/module_common_types.h"
|
||||||
|
#include "webrtc/typedefs.h"
|
||||||
|
|
||||||
|
namespace webrtc {
|
||||||
|
class NewMixHistory;
|
||||||
|
|
||||||
|
// A callback class that all mixer participants must inherit from/implement.
|
||||||
|
class MixerAudioSource {
|
||||||
|
public:
|
||||||
|
// The implementation of this function should update audioFrame with new
|
||||||
|
// audio every time it's called.
|
||||||
|
//
|
||||||
|
// If it returns -1, the frame will not be added to the mix.
|
||||||
|
//
|
||||||
|
// NOTE: This function should not be called. It will remain for a short
|
||||||
|
// time so that subclasses can override it without getting warnings.
|
||||||
|
// TODO(henrik.lundin) Remove this function.
|
||||||
|
virtual int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
|
||||||
|
RTC_CHECK(false);
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
|
||||||
|
// The implementation of GetAudioFrameWithMuted should update audio_frame
|
||||||
|
// with new audio every time it's called. The return value will be
|
||||||
|
// interpreted as follows.
|
||||||
|
enum class AudioFrameInfo {
|
||||||
|
kNormal, // The samples in audio_frame are valid and should be used.
|
||||||
|
kMuted, // The samples in audio_frame should not be used, but should be
|
||||||
|
// implicitly interpreted as zero. Other fields in audio_frame
|
||||||
|
// may be read and should contain meaningful values.
|
||||||
|
kError // audio_frame will not be used.
|
||||||
|
};
|
||||||
|
|
||||||
|
virtual AudioFrameInfo GetAudioFrameWithMuted(int32_t id,
|
||||||
|
AudioFrame* audio_frame) {
|
||||||
|
return GetAudioFrame(id, audio_frame) == -1 ? AudioFrameInfo::kError
|
||||||
|
: AudioFrameInfo::kNormal;
|
||||||
|
}
|
||||||
|
|
||||||
|
// Returns true if the participant was mixed this mix iteration.
|
||||||
|
bool IsMixed() const;
|
||||||
|
|
||||||
|
// This function specifies the sampling frequency needed for the AudioFrame
|
||||||
|
// for future GetAudioFrame(..) calls.
|
||||||
|
virtual int32_t NeededFrequency(int32_t id) const = 0;
|
||||||
|
|
||||||
|
NewMixHistory* _mixHistory;
|
||||||
|
|
||||||
|
protected:
|
||||||
|
MixerAudioSource();
|
||||||
|
virtual ~MixerAudioSource();
|
||||||
|
};
|
||||||
|
|
||||||
|
class OldAudioMixerOutputReceiver {
|
||||||
|
public:
|
||||||
|
// This callback function provides the mixed audio for this mix iteration.
|
||||||
|
// Note that uniqueAudioFrames is an array of AudioFrame pointers with the
|
||||||
|
// size according to the size parameter.
|
||||||
|
virtual void NewMixedAudio(const int32_t id,
|
||||||
|
const AudioFrame& generalAudioFrame,
|
||||||
|
const AudioFrame** uniqueAudioFrames,
|
||||||
|
const uint32_t size) = 0;
|
||||||
|
|
||||||
|
protected:
|
||||||
|
OldAudioMixerOutputReceiver() {}
|
||||||
|
virtual ~OldAudioMixerOutputReceiver() {}
|
||||||
|
};
|
||||||
|
} // namespace webrtc
|
||||||
|
|
||||||
|
#endif // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_
|
||||||
@ -0,0 +1,74 @@
|
|||||||
|
/*
|
||||||
|
* Copyright (c) 2011 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 WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_NEW_AUDIO_CONFERENCE_MIXER_H_
|
||||||
|
#define WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_NEW_AUDIO_CONFERENCE_MIXER_H_
|
||||||
|
|
||||||
|
#include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
|
||||||
|
#include "webrtc/modules/include/module.h"
|
||||||
|
#include "webrtc/modules/include/module_common_types.h"
|
||||||
|
|
||||||
|
namespace webrtc {
|
||||||
|
class OldAudioMixerOutputReceiver;
|
||||||
|
class MixerAudioSource;
|
||||||
|
class Trace;
|
||||||
|
|
||||||
|
class NewAudioConferenceMixer : public Module {
|
||||||
|
public:
|
||||||
|
enum { kMaximumAmountOfMixedParticipants = 3 };
|
||||||
|
enum Frequency {
|
||||||
|
kNbInHz = 8000,
|
||||||
|
kWbInHz = 16000,
|
||||||
|
kSwbInHz = 32000,
|
||||||
|
kFbInHz = 48000,
|
||||||
|
kLowestPossible = -1,
|
||||||
|
kDefaultFrequency = kWbInHz
|
||||||
|
};
|
||||||
|
|
||||||
|
// Factory method. Constructor disabled.
|
||||||
|
static NewAudioConferenceMixer* Create(int id);
|
||||||
|
virtual ~NewAudioConferenceMixer() {}
|
||||||
|
|
||||||
|
// Module functions
|
||||||
|
int64_t TimeUntilNextProcess() override = 0;
|
||||||
|
void Process() override = 0;
|
||||||
|
|
||||||
|
// Register/unregister a callback class for receiving the mixed audio.
|
||||||
|
virtual int32_t RegisterMixedStreamCallback(
|
||||||
|
OldAudioMixerOutputReceiver* receiver) = 0;
|
||||||
|
virtual int32_t UnRegisterMixedStreamCallback() = 0;
|
||||||
|
|
||||||
|
// Add/remove participants as candidates for mixing.
|
||||||
|
virtual int32_t SetMixabilityStatus(MixerAudioSource* participant,
|
||||||
|
bool mixable) = 0;
|
||||||
|
// Returns true if a participant is a candidate for mixing.
|
||||||
|
virtual bool MixabilityStatus(const MixerAudioSource& participant) const = 0;
|
||||||
|
|
||||||
|
// Inform the mixer that the participant should always be mixed and not
|
||||||
|
// count toward the number of mixed participants. Note that a participant
|
||||||
|
// must have been added to the mixer (by calling SetMixabilityStatus())
|
||||||
|
// before this function can be successfully called.
|
||||||
|
virtual int32_t SetAnonymousMixabilityStatus(MixerAudioSource* participant,
|
||||||
|
bool mixable) = 0;
|
||||||
|
// Returns true if the participant is mixed anonymously.
|
||||||
|
virtual bool AnonymousMixabilityStatus(
|
||||||
|
const MixerAudioSource& participant) const = 0;
|
||||||
|
|
||||||
|
// Set the minimum sampling frequency at which to mix. The mixing algorithm
|
||||||
|
// may still choose to mix at a higher samling frequency to avoid
|
||||||
|
// downsampling of audio contributing to the mixed audio.
|
||||||
|
virtual int32_t SetMinimumMixingFrequency(Frequency freq) = 0;
|
||||||
|
|
||||||
|
protected:
|
||||||
|
NewAudioConferenceMixer() {}
|
||||||
|
};
|
||||||
|
} // namespace webrtc
|
||||||
|
|
||||||
|
#endif // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_NEW_AUDIO_CONFERENCE_MIXER_H_
|
||||||
5
webrtc/modules/audio_mixer/source/OWNERS
Normal file
5
webrtc/modules/audio_mixer/source/OWNERS
Normal file
@ -0,0 +1,5 @@
|
|||||||
|
|
||||||
|
# These are for the common case of adding or renaming files. If you're doing
|
||||||
|
# structural changes, please get a review from a reviewer in this file.
|
||||||
|
per-file *.gyp=*
|
||||||
|
per-file *.gypi=*
|
||||||
@ -0,0 +1,898 @@
|
|||||||
|
/*
|
||||||
|
* 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.
|
||||||
|
*/
|
||||||
|
|
||||||
|
#include "webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h"
|
||||||
|
|
||||||
|
#include <algorithm>
|
||||||
|
|
||||||
|
#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
|
||||||
|
#include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
|
||||||
|
#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
||||||
|
#include "webrtc/modules/utility/include/audio_frame_operations.h"
|
||||||
|
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
|
||||||
|
#include "webrtc/system_wrappers/include/trace.h"
|
||||||
|
|
||||||
|
namespace webrtc {
|
||||||
|
namespace {
|
||||||
|
|
||||||
|
struct ParticipantFrameStruct {
|
||||||
|
ParticipantFrameStruct(MixerAudioSource* p, AudioFrame* a, bool m)
|
||||||
|
: participant(p), audioFrame(a), muted(m) {}
|
||||||
|
MixerAudioSource* participant;
|
||||||
|
AudioFrame* audioFrame;
|
||||||
|
bool muted;
|
||||||
|
};
|
||||||
|
|
||||||
|
typedef std::list<ParticipantFrameStruct*> ParticipantFrameStructList;
|
||||||
|
|
||||||
|
// Mix |frame| into |mixed_frame|, with saturation protection and upmixing.
|
||||||
|
// These effects are applied to |frame| itself prior to mixing. Assumes that
|
||||||
|
// |mixed_frame| always has at least as many channels as |frame|. Supports
|
||||||
|
// stereo at most.
|
||||||
|
//
|
||||||
|
// TODO(andrew): consider not modifying |frame| here.
|
||||||
|
void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
|
||||||
|
assert(mixed_frame->num_channels_ >= frame->num_channels_);
|
||||||
|
if (use_limiter) {
|
||||||
|
// Divide by two to avoid saturation in the mixing.
|
||||||
|
// This is only meaningful if the limiter will be used.
|
||||||
|
*frame >>= 1;
|
||||||
|
}
|
||||||
|
if (mixed_frame->num_channels_ > frame->num_channels_) {
|
||||||
|
// We only support mono-to-stereo.
|
||||||
|
assert(mixed_frame->num_channels_ == 2 && frame->num_channels_ == 1);
|
||||||
|
AudioFrameOperations::MonoToStereo(frame);
|
||||||
|
}
|
||||||
|
|
||||||
|
*mixed_frame += *frame;
|
||||||
|
}
|
||||||
|
|
||||||
|
// Return the max number of channels from a |list| composed of AudioFrames.
|
||||||
|
size_t MaxNumChannels(const AudioFrameList* list) {
|
||||||
|
size_t max_num_channels = 1;
|
||||||
|
for (AudioFrameList::const_iterator iter = list->begin(); iter != list->end();
|
||||||
|
++iter) {
|
||||||
|
max_num_channels = std::max(max_num_channels, (*iter).frame->num_channels_);
|
||||||
|
}
|
||||||
|
return max_num_channels;
|
||||||
|
}
|
||||||
|
|
||||||
|
} // namespace
|
||||||
|
|
||||||
|
MixerAudioSource::MixerAudioSource() : _mixHistory(new NewMixHistory()) {}
|
||||||
|
|
||||||
|
MixerAudioSource::~MixerAudioSource() {
|
||||||
|
delete _mixHistory;
|
||||||
|
}
|
||||||
|
|
||||||
|
bool MixerAudioSource::IsMixed() const {
|
||||||
|
return _mixHistory->IsMixed();
|
||||||
|
}
|
||||||
|
|
||||||
|
NewMixHistory::NewMixHistory() : _isMixed(0) {}
|
||||||
|
|
||||||
|
NewMixHistory::~NewMixHistory() {}
|
||||||
|
|
||||||
|
bool NewMixHistory::IsMixed() const {
|
||||||
|
return _isMixed;
|
||||||
|
}
|
||||||
|
|
||||||
|
bool NewMixHistory::WasMixed() const {
|
||||||
|
// Was mixed is the same as is mixed depending on perspective. This function
|
||||||
|
// is for the perspective of NewAudioConferenceMixerImpl.
|
||||||
|
return IsMixed();
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t NewMixHistory::SetIsMixed(const bool mixed) {
|
||||||
|
_isMixed = mixed;
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
void NewMixHistory::ResetMixedStatus() {
|
||||||
|
_isMixed = false;
|
||||||
|
}
|
||||||
|
|
||||||
|
NewAudioConferenceMixer* NewAudioConferenceMixer::Create(int id) {
|
||||||
|
NewAudioConferenceMixerImpl* mixer = new NewAudioConferenceMixerImpl(id);
|
||||||
|
if (!mixer->Init()) {
|
||||||
|
delete mixer;
|
||||||
|
return NULL;
|
||||||
|
}
|
||||||
|
return mixer;
|
||||||
|
}
|
||||||
|
|
||||||
|
NewAudioConferenceMixerImpl::NewAudioConferenceMixerImpl(int id)
|
||||||
|
: _id(id),
|
||||||
|
_minimumMixingFreq(kLowestPossible),
|
||||||
|
_mixReceiver(NULL),
|
||||||
|
_outputFrequency(kDefaultFrequency),
|
||||||
|
_sampleSize(0),
|
||||||
|
_audioFramePool(NULL),
|
||||||
|
_participantList(),
|
||||||
|
_additionalParticipantList(),
|
||||||
|
_numMixedParticipants(0),
|
||||||
|
use_limiter_(true),
|
||||||
|
_timeStamp(0),
|
||||||
|
_timeScheduler(kProcessPeriodicityInMs),
|
||||||
|
_processCalls(0) {}
|
||||||
|
|
||||||
|
bool NewAudioConferenceMixerImpl::Init() {
|
||||||
|
_crit.reset(CriticalSectionWrapper::CreateCriticalSection());
|
||||||
|
if (_crit.get() == NULL)
|
||||||
|
return false;
|
||||||
|
|
||||||
|
_cbCrit.reset(CriticalSectionWrapper::CreateCriticalSection());
|
||||||
|
if (_cbCrit.get() == NULL)
|
||||||
|
return false;
|
||||||
|
|
||||||
|
Config config;
|
||||||
|
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
|
||||||
|
_limiter.reset(AudioProcessing::Create(config));
|
||||||
|
if (!_limiter.get())
|
||||||
|
return false;
|
||||||
|
|
||||||
|
MemoryPool<AudioFrame>::CreateMemoryPool(_audioFramePool,
|
||||||
|
DEFAULT_AUDIO_FRAME_POOLSIZE);
|
||||||
|
if (_audioFramePool == NULL)
|
||||||
|
return false;
|
||||||
|
|
||||||
|
if (SetOutputFrequency(kDefaultFrequency) == -1)
|
||||||
|
return false;
|
||||||
|
|
||||||
|
if (_limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
|
||||||
|
_limiter->kNoError)
|
||||||
|
return false;
|
||||||
|
|
||||||
|
// We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
|
||||||
|
// divide-by-2 but -7 is used instead to give a bit of headroom since the
|
||||||
|
// AGC is not a hard limiter.
|
||||||
|
if (_limiter->gain_control()->set_target_level_dbfs(7) != _limiter->kNoError)
|
||||||
|
return false;
|
||||||
|
|
||||||
|
if (_limiter->gain_control()->set_compression_gain_db(0) !=
|
||||||
|
_limiter->kNoError)
|
||||||
|
return false;
|
||||||
|
|
||||||
|
if (_limiter->gain_control()->enable_limiter(true) != _limiter->kNoError)
|
||||||
|
return false;
|
||||||
|
|
||||||
|
if (_limiter->gain_control()->Enable(true) != _limiter->kNoError)
|
||||||
|
return false;
|
||||||
|
|
||||||
|
return true;
|
||||||
|
}
|
||||||
|
|
||||||
|
NewAudioConferenceMixerImpl::~NewAudioConferenceMixerImpl() {
|
||||||
|
MemoryPool<AudioFrame>::DeleteMemoryPool(_audioFramePool);
|
||||||
|
assert(_audioFramePool == NULL);
|
||||||
|
}
|
||||||
|
|
||||||
|
// Process should be called every kProcessPeriodicityInMs ms
|
||||||
|
int64_t NewAudioConferenceMixerImpl::TimeUntilNextProcess() {
|
||||||
|
int64_t timeUntilNextProcess = 0;
|
||||||
|
CriticalSectionScoped cs(_crit.get());
|
||||||
|
if (_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0) {
|
||||||
|
WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
|
||||||
|
"failed in TimeToNextUpdate() call");
|
||||||
|
// Sanity check
|
||||||
|
assert(false);
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
return timeUntilNextProcess;
|
||||||
|
}
|
||||||
|
|
||||||
|
void NewAudioConferenceMixerImpl::Process() {
|
||||||
|
size_t remainingParticipantsAllowedToMix = kMaximumAmountOfMixedParticipants;
|
||||||
|
{
|
||||||
|
CriticalSectionScoped cs(_crit.get());
|
||||||
|
assert(_processCalls == 0);
|
||||||
|
_processCalls++;
|
||||||
|
|
||||||
|
// Let the scheduler know that we are running one iteration.
|
||||||
|
_timeScheduler.UpdateScheduler();
|
||||||
|
}
|
||||||
|
|
||||||
|
AudioFrameList mixList;
|
||||||
|
AudioFrameList rampOutList;
|
||||||
|
AudioFrameList additionalFramesList;
|
||||||
|
std::map<int, MixerAudioSource*> mixedParticipantsMap;
|
||||||
|
{
|
||||||
|
CriticalSectionScoped cs(_cbCrit.get());
|
||||||
|
|
||||||
|
int32_t lowFreq = GetLowestMixingFrequency();
|
||||||
|
// SILK can run in 12 kHz and 24 kHz. These frequencies are not
|
||||||
|
// supported so use the closest higher frequency to not lose any
|
||||||
|
// information.
|
||||||
|
// TODO(henrike): this is probably more appropriate to do in
|
||||||
|
// GetLowestMixingFrequency().
|
||||||
|
if (lowFreq == 12000) {
|
||||||
|
lowFreq = 16000;
|
||||||
|
} else if (lowFreq == 24000) {
|
||||||
|
lowFreq = 32000;
|
||||||
|
}
|
||||||
|
if (lowFreq <= 0) {
|
||||||
|
CriticalSectionScoped cs(_crit.get());
|
||||||
|
_processCalls--;
|
||||||
|
return;
|
||||||
|
} else {
|
||||||
|
switch (lowFreq) {
|
||||||
|
case 8000:
|
||||||
|
if (OutputFrequency() != kNbInHz) {
|
||||||
|
SetOutputFrequency(kNbInHz);
|
||||||
|
}
|
||||||
|
break;
|
||||||
|
case 16000:
|
||||||
|
if (OutputFrequency() != kWbInHz) {
|
||||||
|
SetOutputFrequency(kWbInHz);
|
||||||
|
}
|
||||||
|
break;
|
||||||
|
case 32000:
|
||||||
|
if (OutputFrequency() != kSwbInHz) {
|
||||||
|
SetOutputFrequency(kSwbInHz);
|
||||||
|
}
|
||||||
|
break;
|
||||||
|
case 48000:
|
||||||
|
if (OutputFrequency() != kFbInHz) {
|
||||||
|
SetOutputFrequency(kFbInHz);
|
||||||
|
}
|
||||||
|
break;
|
||||||
|
default:
|
||||||
|
assert(false);
|
||||||
|
|
||||||
|
CriticalSectionScoped cs(_crit.get());
|
||||||
|
_processCalls--;
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
UpdateToMix(&mixList, &rampOutList, &mixedParticipantsMap,
|
||||||
|
&remainingParticipantsAllowedToMix);
|
||||||
|
|
||||||
|
GetAdditionalAudio(&additionalFramesList);
|
||||||
|
UpdateMixedStatus(mixedParticipantsMap);
|
||||||
|
}
|
||||||
|
|
||||||
|
// Get an AudioFrame for mixing from the memory pool.
|
||||||
|
AudioFrame* mixedAudio = NULL;
|
||||||
|
if (_audioFramePool->PopMemory(mixedAudio) == -1) {
|
||||||
|
WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
|
||||||
|
"failed PopMemory() call");
|
||||||
|
assert(false);
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
|
||||||
|
{
|
||||||
|
CriticalSectionScoped cs(_crit.get());
|
||||||
|
|
||||||
|
// TODO(henrike): it might be better to decide the number of channels
|
||||||
|
// with an API instead of dynamically.
|
||||||
|
|
||||||
|
// Find the max channels over all mixing lists.
|
||||||
|
const size_t num_mixed_channels =
|
||||||
|
std::max(MaxNumChannels(&mixList),
|
||||||
|
std::max(MaxNumChannels(&additionalFramesList),
|
||||||
|
MaxNumChannels(&rampOutList)));
|
||||||
|
|
||||||
|
mixedAudio->UpdateFrame(-1, _timeStamp, NULL, 0, _outputFrequency,
|
||||||
|
AudioFrame::kNormalSpeech, AudioFrame::kVadPassive,
|
||||||
|
num_mixed_channels);
|
||||||
|
|
||||||
|
_timeStamp += static_cast<uint32_t>(_sampleSize);
|
||||||
|
|
||||||
|
// We only use the limiter if it supports the output sample rate and
|
||||||
|
// we're actually mixing multiple streams.
|
||||||
|
use_limiter_ = _numMixedParticipants > 1 &&
|
||||||
|
_outputFrequency <= AudioProcessing::kMaxNativeSampleRateHz;
|
||||||
|
|
||||||
|
MixFromList(mixedAudio, mixList);
|
||||||
|
MixAnonomouslyFromList(mixedAudio, additionalFramesList);
|
||||||
|
MixAnonomouslyFromList(mixedAudio, rampOutList);
|
||||||
|
|
||||||
|
if (mixedAudio->samples_per_channel_ == 0) {
|
||||||
|
// Nothing was mixed, set the audio samples to silence.
|
||||||
|
mixedAudio->samples_per_channel_ = _sampleSize;
|
||||||
|
mixedAudio->Mute();
|
||||||
|
} else {
|
||||||
|
// Only call the limiter if we have something to mix.
|
||||||
|
LimitMixedAudio(mixedAudio);
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
{
|
||||||
|
CriticalSectionScoped cs(_cbCrit.get());
|
||||||
|
if (_mixReceiver != NULL) {
|
||||||
|
const AudioFrame** dummy = NULL;
|
||||||
|
_mixReceiver->NewMixedAudio(_id, *mixedAudio, dummy, 0);
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
// Reclaim all outstanding memory.
|
||||||
|
_audioFramePool->PushMemory(mixedAudio);
|
||||||
|
ClearAudioFrameList(&mixList);
|
||||||
|
ClearAudioFrameList(&rampOutList);
|
||||||
|
ClearAudioFrameList(&additionalFramesList);
|
||||||
|
{
|
||||||
|
CriticalSectionScoped cs(_crit.get());
|
||||||
|
_processCalls--;
|
||||||
|
}
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t NewAudioConferenceMixerImpl::RegisterMixedStreamCallback(
|
||||||
|
OldAudioMixerOutputReceiver* mixReceiver) {
|
||||||
|
CriticalSectionScoped cs(_cbCrit.get());
|
||||||
|
if (_mixReceiver != NULL) {
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
_mixReceiver = mixReceiver;
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t NewAudioConferenceMixerImpl::UnRegisterMixedStreamCallback() {
|
||||||
|
CriticalSectionScoped cs(_cbCrit.get());
|
||||||
|
if (_mixReceiver == NULL) {
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
_mixReceiver = NULL;
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t NewAudioConferenceMixerImpl::SetOutputFrequency(
|
||||||
|
const Frequency& frequency) {
|
||||||
|
CriticalSectionScoped cs(_crit.get());
|
||||||
|
|
||||||
|
_outputFrequency = frequency;
|
||||||
|
_sampleSize =
|
||||||
|
static_cast<size_t>((_outputFrequency * kProcessPeriodicityInMs) / 1000);
|
||||||
|
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
NewAudioConferenceMixer::Frequency
|
||||||
|
NewAudioConferenceMixerImpl::OutputFrequency() const {
|
||||||
|
CriticalSectionScoped cs(_crit.get());
|
||||||
|
return _outputFrequency;
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t NewAudioConferenceMixerImpl::SetMixabilityStatus(
|
||||||
|
MixerAudioSource* participant,
|
||||||
|
bool mixable) {
|
||||||
|
if (!mixable) {
|
||||||
|
// Anonymous participants are in a separate list. Make sure that the
|
||||||
|
// participant is in the _participantList if it is being mixed.
|
||||||
|
SetAnonymousMixabilityStatus(participant, false);
|
||||||
|
}
|
||||||
|
size_t numMixedParticipants;
|
||||||
|
{
|
||||||
|
CriticalSectionScoped cs(_cbCrit.get());
|
||||||
|
const bool isMixed = IsParticipantInList(*participant, _participantList);
|
||||||
|
// API must be called with a new state.
|
||||||
|
if (!(mixable ^ isMixed)) {
|
||||||
|
WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
|
||||||
|
"Mixable is aready %s", isMixed ? "ON" : "off");
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
bool success = false;
|
||||||
|
if (mixable) {
|
||||||
|
success = AddParticipantToList(participant, &_participantList);
|
||||||
|
} else {
|
||||||
|
success = RemoveParticipantFromList(participant, &_participantList);
|
||||||
|
}
|
||||||
|
if (!success) {
|
||||||
|
WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
|
||||||
|
"failed to %s participant", mixable ? "add" : "remove");
|
||||||
|
assert(false);
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
|
||||||
|
size_t numMixedNonAnonymous = _participantList.size();
|
||||||
|
if (numMixedNonAnonymous > kMaximumAmountOfMixedParticipants) {
|
||||||
|
numMixedNonAnonymous = kMaximumAmountOfMixedParticipants;
|
||||||
|
}
|
||||||
|
numMixedParticipants =
|
||||||
|
numMixedNonAnonymous + _additionalParticipantList.size();
|
||||||
|
}
|
||||||
|
// A MixerAudioSource was added or removed. Make sure the scratch
|
||||||
|
// buffer is updated if necessary.
|
||||||
|
// Note: The scratch buffer may only be updated in Process().
|
||||||
|
CriticalSectionScoped cs(_crit.get());
|
||||||
|
_numMixedParticipants = numMixedParticipants;
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
bool NewAudioConferenceMixerImpl::MixabilityStatus(
|
||||||
|
const MixerAudioSource& participant) const {
|
||||||
|
CriticalSectionScoped cs(_cbCrit.get());
|
||||||
|
return IsParticipantInList(participant, _participantList);
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t NewAudioConferenceMixerImpl::SetAnonymousMixabilityStatus(
|
||||||
|
MixerAudioSource* participant,
|
||||||
|
bool anonymous) {
|
||||||
|
CriticalSectionScoped cs(_cbCrit.get());
|
||||||
|
if (IsParticipantInList(*participant, _additionalParticipantList)) {
|
||||||
|
if (anonymous) {
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
if (!RemoveParticipantFromList(participant, &_additionalParticipantList)) {
|
||||||
|
WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
|
||||||
|
"unable to remove participant from anonymous list");
|
||||||
|
assert(false);
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
return AddParticipantToList(participant, &_participantList) ? 0 : -1;
|
||||||
|
}
|
||||||
|
if (!anonymous) {
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
const bool mixable =
|
||||||
|
RemoveParticipantFromList(participant, &_participantList);
|
||||||
|
if (!mixable) {
|
||||||
|
WEBRTC_TRACE(
|
||||||
|
kTraceWarning, kTraceAudioMixerServer, _id,
|
||||||
|
"participant must be registered before turning it into anonymous");
|
||||||
|
// Setting anonymous status is only possible if MixerAudioSource is
|
||||||
|
// already registered.
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
return AddParticipantToList(participant, &_additionalParticipantList) ? 0
|
||||||
|
: -1;
|
||||||
|
}
|
||||||
|
|
||||||
|
bool NewAudioConferenceMixerImpl::AnonymousMixabilityStatus(
|
||||||
|
const MixerAudioSource& participant) const {
|
||||||
|
CriticalSectionScoped cs(_cbCrit.get());
|
||||||
|
return IsParticipantInList(participant, _additionalParticipantList);
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t NewAudioConferenceMixerImpl::SetMinimumMixingFrequency(Frequency freq) {
|
||||||
|
// Make sure that only allowed sampling frequencies are used. Use closest
|
||||||
|
// higher sampling frequency to avoid losing information.
|
||||||
|
if (static_cast<int>(freq) == 12000) {
|
||||||
|
freq = kWbInHz;
|
||||||
|
} else if (static_cast<int>(freq) == 24000) {
|
||||||
|
freq = kSwbInHz;
|
||||||
|
}
|
||||||
|
|
||||||
|
if ((freq == kNbInHz) || (freq == kWbInHz) || (freq == kSwbInHz) ||
|
||||||
|
(freq == kLowestPossible)) {
|
||||||
|
_minimumMixingFreq = freq;
|
||||||
|
return 0;
|
||||||
|
} else {
|
||||||
|
WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
|
||||||
|
"SetMinimumMixingFrequency incorrect frequency: %i", freq);
|
||||||
|
assert(false);
|
||||||
|
return -1;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
// Check all AudioFrames that are to be mixed. The highest sampling frequency
|
||||||
|
// found is the lowest that can be used without losing information.
|
||||||
|
int32_t NewAudioConferenceMixerImpl::GetLowestMixingFrequency() const {
|
||||||
|
const int participantListFrequency =
|
||||||
|
GetLowestMixingFrequencyFromList(_participantList);
|
||||||
|
const int anonymousListFrequency =
|
||||||
|
GetLowestMixingFrequencyFromList(_additionalParticipantList);
|
||||||
|
const int highestFreq = (participantListFrequency > anonymousListFrequency)
|
||||||
|
? participantListFrequency
|
||||||
|
: anonymousListFrequency;
|
||||||
|
// Check if the user specified a lowest mixing frequency.
|
||||||
|
if (_minimumMixingFreq != kLowestPossible) {
|
||||||
|
if (_minimumMixingFreq > highestFreq) {
|
||||||
|
return _minimumMixingFreq;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
return highestFreq;
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t NewAudioConferenceMixerImpl::GetLowestMixingFrequencyFromList(
|
||||||
|
const MixerAudioSourceList& mixList) const {
|
||||||
|
int32_t highestFreq = 8000;
|
||||||
|
for (MixerAudioSourceList::const_iterator iter = mixList.begin();
|
||||||
|
iter != mixList.end(); ++iter) {
|
||||||
|
const int32_t neededFrequency = (*iter)->NeededFrequency(_id);
|
||||||
|
if (neededFrequency > highestFreq) {
|
||||||
|
highestFreq = neededFrequency;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
return highestFreq;
|
||||||
|
}
|
||||||
|
|
||||||
|
void NewAudioConferenceMixerImpl::UpdateToMix(
|
||||||
|
AudioFrameList* mixList,
|
||||||
|
AudioFrameList* rampOutList,
|
||||||
|
std::map<int, MixerAudioSource*>* mixParticipantList,
|
||||||
|
size_t* maxAudioFrameCounter) const {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
|
||||||
|
"UpdateToMix(mixList,rampOutList,mixParticipantList,%d)",
|
||||||
|
*maxAudioFrameCounter);
|
||||||
|
const size_t mixListStartSize = mixList->size();
|
||||||
|
AudioFrameList activeList;
|
||||||
|
// Struct needed by the passive lists to keep track of which AudioFrame
|
||||||
|
// belongs to which MixerAudioSource.
|
||||||
|
ParticipantFrameStructList passiveWasNotMixedList;
|
||||||
|
ParticipantFrameStructList passiveWasMixedList;
|
||||||
|
for (MixerAudioSourceList::const_iterator participant =
|
||||||
|
_participantList.begin();
|
||||||
|
participant != _participantList.end(); ++participant) {
|
||||||
|
// Stop keeping track of passive participants if there are already
|
||||||
|
// enough participants available (they wont be mixed anyway).
|
||||||
|
bool mustAddToPassiveList =
|
||||||
|
(*maxAudioFrameCounter >
|
||||||
|
(activeList.size() + passiveWasMixedList.size() +
|
||||||
|
passiveWasNotMixedList.size()));
|
||||||
|
|
||||||
|
bool wasMixed = false;
|
||||||
|
wasMixed = (*participant)->_mixHistory->WasMixed();
|
||||||
|
AudioFrame* audioFrame = NULL;
|
||||||
|
if (_audioFramePool->PopMemory(audioFrame) == -1) {
|
||||||
|
WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
|
||||||
|
"failed PopMemory() call");
|
||||||
|
assert(false);
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
audioFrame->sample_rate_hz_ = _outputFrequency;
|
||||||
|
|
||||||
|
auto ret = (*participant)->GetAudioFrameWithMuted(_id, audioFrame);
|
||||||
|
if (ret == MixerAudioSource::AudioFrameInfo::kError) {
|
||||||
|
WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
|
||||||
|
"failed to GetAudioFrameWithMuted() from participant");
|
||||||
|
_audioFramePool->PushMemory(audioFrame);
|
||||||
|
continue;
|
||||||
|
}
|
||||||
|
const bool muted = (ret == MixerAudioSource::AudioFrameInfo::kMuted);
|
||||||
|
if (_participantList.size() != 1) {
|
||||||
|
// TODO(wu): Issue 3390, add support for multiple participants case.
|
||||||
|
audioFrame->ntp_time_ms_ = -1;
|
||||||
|
}
|
||||||
|
|
||||||
|
// TODO(henrike): this assert triggers in some test cases where SRTP is
|
||||||
|
// used which prevents NetEQ from making a VAD. Temporarily disable this
|
||||||
|
// assert until the problem is fixed on a higher level.
|
||||||
|
// assert(audioFrame->vad_activity_ != AudioFrame::kVadUnknown);
|
||||||
|
if (audioFrame->vad_activity_ == AudioFrame::kVadUnknown) {
|
||||||
|
WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
|
||||||
|
"invalid VAD state from participant");
|
||||||
|
}
|
||||||
|
|
||||||
|
if (audioFrame->vad_activity_ == AudioFrame::kVadActive) {
|
||||||
|
if (!wasMixed && !muted) {
|
||||||
|
RampIn(*audioFrame);
|
||||||
|
}
|
||||||
|
|
||||||
|
if (activeList.size() >= *maxAudioFrameCounter) {
|
||||||
|
// There are already more active participants than should be
|
||||||
|
// mixed. Only keep the ones with the highest energy.
|
||||||
|
AudioFrameList::iterator replaceItem;
|
||||||
|
uint32_t lowestEnergy = muted ? 0 : CalculateEnergy(*audioFrame);
|
||||||
|
|
||||||
|
bool found_replace_item = false;
|
||||||
|
for (AudioFrameList::iterator iter = activeList.begin();
|
||||||
|
iter != activeList.end(); ++iter) {
|
||||||
|
const uint32_t energy = muted ? 0 : CalculateEnergy(*iter->frame);
|
||||||
|
if (energy < lowestEnergy) {
|
||||||
|
replaceItem = iter;
|
||||||
|
lowestEnergy = energy;
|
||||||
|
found_replace_item = true;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
if (found_replace_item) {
|
||||||
|
RTC_DCHECK(!muted); // Cannot replace with a muted frame.
|
||||||
|
FrameAndMuteInfo replaceFrame = *replaceItem;
|
||||||
|
|
||||||
|
bool replaceWasMixed = false;
|
||||||
|
std::map<int, MixerAudioSource*>::const_iterator it =
|
||||||
|
mixParticipantList->find(replaceFrame.frame->id_);
|
||||||
|
|
||||||
|
// When a frame is pushed to |activeList| it is also pushed
|
||||||
|
// to mixParticipantList with the frame's id. This means
|
||||||
|
// that the Find call above should never fail.
|
||||||
|
assert(it != mixParticipantList->end());
|
||||||
|
replaceWasMixed = it->second->_mixHistory->WasMixed();
|
||||||
|
|
||||||
|
mixParticipantList->erase(replaceFrame.frame->id_);
|
||||||
|
activeList.erase(replaceItem);
|
||||||
|
|
||||||
|
activeList.push_front(FrameAndMuteInfo(audioFrame, muted));
|
||||||
|
(*mixParticipantList)[audioFrame->id_] = *participant;
|
||||||
|
assert(mixParticipantList->size() <=
|
||||||
|
kMaximumAmountOfMixedParticipants);
|
||||||
|
|
||||||
|
if (replaceWasMixed) {
|
||||||
|
if (!replaceFrame.muted) {
|
||||||
|
RampOut(*replaceFrame.frame);
|
||||||
|
}
|
||||||
|
rampOutList->push_back(replaceFrame);
|
||||||
|
assert(rampOutList->size() <= kMaximumAmountOfMixedParticipants);
|
||||||
|
} else {
|
||||||
|
_audioFramePool->PushMemory(replaceFrame.frame);
|
||||||
|
}
|
||||||
|
} else {
|
||||||
|
if (wasMixed) {
|
||||||
|
if (!muted) {
|
||||||
|
RampOut(*audioFrame);
|
||||||
|
}
|
||||||
|
rampOutList->push_back(FrameAndMuteInfo(audioFrame, muted));
|
||||||
|
assert(rampOutList->size() <= kMaximumAmountOfMixedParticipants);
|
||||||
|
} else {
|
||||||
|
_audioFramePool->PushMemory(audioFrame);
|
||||||
|
}
|
||||||
|
}
|
||||||
|
} else {
|
||||||
|
activeList.push_front(FrameAndMuteInfo(audioFrame, muted));
|
||||||
|
(*mixParticipantList)[audioFrame->id_] = *participant;
|
||||||
|
assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants);
|
||||||
|
}
|
||||||
|
} else {
|
||||||
|
if (wasMixed) {
|
||||||
|
ParticipantFrameStruct* part_struct =
|
||||||
|
new ParticipantFrameStruct(*participant, audioFrame, muted);
|
||||||
|
passiveWasMixedList.push_back(part_struct);
|
||||||
|
} else if (mustAddToPassiveList) {
|
||||||
|
if (!muted) {
|
||||||
|
RampIn(*audioFrame);
|
||||||
|
}
|
||||||
|
ParticipantFrameStruct* part_struct =
|
||||||
|
new ParticipantFrameStruct(*participant, audioFrame, muted);
|
||||||
|
passiveWasNotMixedList.push_back(part_struct);
|
||||||
|
} else {
|
||||||
|
_audioFramePool->PushMemory(audioFrame);
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
assert(activeList.size() <= *maxAudioFrameCounter);
|
||||||
|
// At this point it is known which participants should be mixed. Transfer
|
||||||
|
// this information to this functions output parameters.
|
||||||
|
for (AudioFrameList::const_iterator iter = activeList.begin();
|
||||||
|
iter != activeList.end(); ++iter) {
|
||||||
|
mixList->push_back(*iter);
|
||||||
|
}
|
||||||
|
activeList.clear();
|
||||||
|
// Always mix a constant number of AudioFrames. If there aren't enough
|
||||||
|
// active participants mix passive ones. Starting with those that was mixed
|
||||||
|
// last iteration.
|
||||||
|
for (ParticipantFrameStructList::const_iterator iter =
|
||||||
|
passiveWasMixedList.begin();
|
||||||
|
iter != passiveWasMixedList.end(); ++iter) {
|
||||||
|
if (mixList->size() < *maxAudioFrameCounter + mixListStartSize) {
|
||||||
|
mixList->push_back(FrameAndMuteInfo((*iter)->audioFrame, (*iter)->muted));
|
||||||
|
(*mixParticipantList)[(*iter)->audioFrame->id_] = (*iter)->participant;
|
||||||
|
assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants);
|
||||||
|
} else {
|
||||||
|
_audioFramePool->PushMemory((*iter)->audioFrame);
|
||||||
|
}
|
||||||
|
delete *iter;
|
||||||
|
}
|
||||||
|
// And finally the ones that have not been mixed for a while.
|
||||||
|
for (ParticipantFrameStructList::const_iterator iter =
|
||||||
|
passiveWasNotMixedList.begin();
|
||||||
|
iter != passiveWasNotMixedList.end(); ++iter) {
|
||||||
|
if (mixList->size() < *maxAudioFrameCounter + mixListStartSize) {
|
||||||
|
mixList->push_back(FrameAndMuteInfo((*iter)->audioFrame, (*iter)->muted));
|
||||||
|
(*mixParticipantList)[(*iter)->audioFrame->id_] = (*iter)->participant;
|
||||||
|
assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants);
|
||||||
|
} else {
|
||||||
|
_audioFramePool->PushMemory((*iter)->audioFrame);
|
||||||
|
}
|
||||||
|
delete *iter;
|
||||||
|
}
|
||||||
|
assert(*maxAudioFrameCounter + mixListStartSize >= mixList->size());
|
||||||
|
*maxAudioFrameCounter += mixListStartSize - mixList->size();
|
||||||
|
}
|
||||||
|
|
||||||
|
void NewAudioConferenceMixerImpl::GetAdditionalAudio(
|
||||||
|
AudioFrameList* additionalFramesList) const {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
|
||||||
|
"GetAdditionalAudio(additionalFramesList)");
|
||||||
|
// The GetAudioFrameWithMuted() callback may result in the participant being
|
||||||
|
// removed from additionalParticipantList_. If that happens it will
|
||||||
|
// invalidate any iterators. Create a copy of the participants list such
|
||||||
|
// that the list of participants can be traversed safely.
|
||||||
|
MixerAudioSourceList additionalParticipantList;
|
||||||
|
additionalParticipantList.insert(additionalParticipantList.begin(),
|
||||||
|
_additionalParticipantList.begin(),
|
||||||
|
_additionalParticipantList.end());
|
||||||
|
|
||||||
|
for (MixerAudioSourceList::const_iterator participant =
|
||||||
|
additionalParticipantList.begin();
|
||||||
|
participant != additionalParticipantList.end(); ++participant) {
|
||||||
|
AudioFrame* audioFrame = NULL;
|
||||||
|
if (_audioFramePool->PopMemory(audioFrame) == -1) {
|
||||||
|
WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
|
||||||
|
"failed PopMemory() call");
|
||||||
|
assert(false);
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
audioFrame->sample_rate_hz_ = _outputFrequency;
|
||||||
|
auto ret = (*participant)->GetAudioFrameWithMuted(_id, audioFrame);
|
||||||
|
if (ret == MixerAudioSource::AudioFrameInfo::kError) {
|
||||||
|
WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
|
||||||
|
"failed to GetAudioFrameWithMuted() from participant");
|
||||||
|
_audioFramePool->PushMemory(audioFrame);
|
||||||
|
continue;
|
||||||
|
}
|
||||||
|
if (audioFrame->samples_per_channel_ == 0) {
|
||||||
|
// Empty frame. Don't use it.
|
||||||
|
_audioFramePool->PushMemory(audioFrame);
|
||||||
|
continue;
|
||||||
|
}
|
||||||
|
additionalFramesList->push_back(FrameAndMuteInfo(
|
||||||
|
audioFrame, ret == MixerAudioSource::AudioFrameInfo::kMuted));
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
void NewAudioConferenceMixerImpl::UpdateMixedStatus(
|
||||||
|
const std::map<int, MixerAudioSource*>& mixedParticipantsMap) const {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
|
||||||
|
"UpdateMixedStatus(mixedParticipantsMap)");
|
||||||
|
assert(mixedParticipantsMap.size() <= kMaximumAmountOfMixedParticipants);
|
||||||
|
|
||||||
|
// Loop through all participants. If they are in the mix map they
|
||||||
|
// were mixed.
|
||||||
|
for (MixerAudioSourceList::const_iterator participant =
|
||||||
|
_participantList.begin();
|
||||||
|
participant != _participantList.end(); ++participant) {
|
||||||
|
bool isMixed = false;
|
||||||
|
for (std::map<int, MixerAudioSource*>::const_iterator it =
|
||||||
|
mixedParticipantsMap.begin();
|
||||||
|
it != mixedParticipantsMap.end(); ++it) {
|
||||||
|
if (it->second == *participant) {
|
||||||
|
isMixed = true;
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
(*participant)->_mixHistory->SetIsMixed(isMixed);
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
void NewAudioConferenceMixerImpl::ClearAudioFrameList(
|
||||||
|
AudioFrameList* audioFrameList) const {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
|
||||||
|
"ClearAudioFrameList(audioFrameList)");
|
||||||
|
for (AudioFrameList::iterator iter = audioFrameList->begin();
|
||||||
|
iter != audioFrameList->end(); ++iter) {
|
||||||
|
_audioFramePool->PushMemory(iter->frame);
|
||||||
|
}
|
||||||
|
audioFrameList->clear();
|
||||||
|
}
|
||||||
|
|
||||||
|
bool NewAudioConferenceMixerImpl::IsParticipantInList(
|
||||||
|
const MixerAudioSource& participant,
|
||||||
|
const MixerAudioSourceList& participantList) const {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
|
||||||
|
"IsParticipantInList(participant,participantList)");
|
||||||
|
for (MixerAudioSourceList::const_iterator iter = participantList.begin();
|
||||||
|
iter != participantList.end(); ++iter) {
|
||||||
|
if (&participant == *iter) {
|
||||||
|
return true;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
return false;
|
||||||
|
}
|
||||||
|
|
||||||
|
bool NewAudioConferenceMixerImpl::AddParticipantToList(
|
||||||
|
MixerAudioSource* participant,
|
||||||
|
MixerAudioSourceList* participantList) const {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
|
||||||
|
"AddParticipantToList(participant, participantList)");
|
||||||
|
participantList->push_back(participant);
|
||||||
|
// Make sure that the mixed status is correct for new MixerAudioSource.
|
||||||
|
participant->_mixHistory->ResetMixedStatus();
|
||||||
|
return true;
|
||||||
|
}
|
||||||
|
|
||||||
|
bool NewAudioConferenceMixerImpl::RemoveParticipantFromList(
|
||||||
|
MixerAudioSource* participant,
|
||||||
|
MixerAudioSourceList* participantList) const {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
|
||||||
|
"RemoveParticipantFromList(participant, participantList)");
|
||||||
|
for (MixerAudioSourceList::iterator iter = participantList->begin();
|
||||||
|
iter != participantList->end(); ++iter) {
|
||||||
|
if (*iter == participant) {
|
||||||
|
participantList->erase(iter);
|
||||||
|
// Participant is no longer mixed, reset to default.
|
||||||
|
participant->_mixHistory->ResetMixedStatus();
|
||||||
|
return true;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
return false;
|
||||||
|
}
|
||||||
|
|
||||||
|
int32_t NewAudioConferenceMixerImpl::MixFromList(
|
||||||
|
AudioFrame* mixedAudio,
|
||||||
|
const AudioFrameList& audioFrameList) const {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
|
||||||
|
"MixFromList(mixedAudio, audioFrameList)");
|
||||||
|
if (audioFrameList.empty())
|
||||||
|
return 0;
|
||||||
|
|
||||||
|
uint32_t position = 0;
|
||||||
|
|
||||||
|
if (_numMixedParticipants == 1) {
|
||||||
|
mixedAudio->timestamp_ = audioFrameList.front().frame->timestamp_;
|
||||||
|
mixedAudio->elapsed_time_ms_ =
|
||||||
|
audioFrameList.front().frame->elapsed_time_ms_;
|
||||||
|
} else {
|
||||||
|
// TODO(wu): Issue 3390.
|
||||||
|
// Audio frame timestamp is only supported in one channel case.
|
||||||
|
mixedAudio->timestamp_ = 0;
|
||||||
|
mixedAudio->elapsed_time_ms_ = -1;
|
||||||
|
}
|
||||||
|
|
||||||
|
for (AudioFrameList::const_iterator iter = audioFrameList.begin();
|
||||||
|
iter != audioFrameList.end(); ++iter) {
|
||||||
|
if (position >= kMaximumAmountOfMixedParticipants) {
|
||||||
|
WEBRTC_TRACE(
|
||||||
|
kTraceMemory, kTraceAudioMixerServer, _id,
|
||||||
|
"Trying to mix more than max amount of mixed participants:%d!",
|
||||||
|
kMaximumAmountOfMixedParticipants);
|
||||||
|
// Assert and avoid crash
|
||||||
|
assert(false);
|
||||||
|
position = 0;
|
||||||
|
}
|
||||||
|
if (!iter->muted) {
|
||||||
|
MixFrames(mixedAudio, iter->frame, use_limiter_);
|
||||||
|
}
|
||||||
|
|
||||||
|
position++;
|
||||||
|
}
|
||||||
|
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
// TODO(andrew): consolidate this function with MixFromList.
|
||||||
|
int32_t NewAudioConferenceMixerImpl::MixAnonomouslyFromList(
|
||||||
|
AudioFrame* mixedAudio,
|
||||||
|
const AudioFrameList& audioFrameList) const {
|
||||||
|
WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
|
||||||
|
"MixAnonomouslyFromList(mixedAudio, audioFrameList)");
|
||||||
|
|
||||||
|
if (audioFrameList.empty())
|
||||||
|
return 0;
|
||||||
|
|
||||||
|
for (AudioFrameList::const_iterator iter = audioFrameList.begin();
|
||||||
|
iter != audioFrameList.end(); ++iter) {
|
||||||
|
if (!iter->muted) {
|
||||||
|
MixFrames(mixedAudio, iter->frame, use_limiter_);
|
||||||
|
}
|
||||||
|
}
|
||||||
|
return 0;
|
||||||
|
}
|
||||||
|
|
||||||
|
bool NewAudioConferenceMixerImpl::LimitMixedAudio(
|
||||||
|
AudioFrame* mixedAudio) const {
|
||||||
|
if (!use_limiter_) {
|
||||||
|
return true;
|
||||||
|
}
|
||||||
|
|
||||||
|
// Smoothly limit the mixed frame.
|
||||||
|
const int error = _limiter->ProcessStream(mixedAudio);
|
||||||
|
|
||||||
|
// And now we can safely restore the level. This procedure results in
|
||||||
|
// some loss of resolution, deemed acceptable.
|
||||||
|
//
|
||||||
|
// It's possible to apply the gain in the AGC (with a target level of 0 dbFS
|
||||||
|
// and compression gain of 6 dB). However, in the transition frame when this
|
||||||
|
// is enabled (moving from one to two participants) it has the potential to
|
||||||
|
// create discontinuities in the mixed frame.
|
||||||
|
//
|
||||||
|
// Instead we double the frame (with addition since left-shifting a
|
||||||
|
// negative value is undefined).
|
||||||
|
*mixedAudio += *mixedAudio;
|
||||||
|
|
||||||
|
if (error != _limiter->kNoError) {
|
||||||
|
WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
|
||||||
|
"Error from AudioProcessing: %d", error);
|
||||||
|
assert(false);
|
||||||
|
return false;
|
||||||
|
}
|
||||||
|
return true;
|
||||||
|
}
|
||||||
|
} // namespace webrtc
|
||||||
@ -0,0 +1,188 @@
|
|||||||
|
/*
|
||||||
|
* 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 WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
|
||||||
|
#define WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
|
||||||
|
|
||||||
|
#include <list>
|
||||||
|
#include <map>
|
||||||
|
#include <memory>
|
||||||
|
|
||||||
|
#include "webrtc/engine_configurations.h"
|
||||||
|
#include "webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h"
|
||||||
|
#include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
|
||||||
|
#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
|
||||||
|
#include "webrtc/modules/include/module_common_types.h"
|
||||||
|
|
||||||
|
namespace webrtc {
|
||||||
|
class AudioProcessing;
|
||||||
|
class CriticalSectionWrapper;
|
||||||
|
|
||||||
|
struct FrameAndMuteInfo {
|
||||||
|
FrameAndMuteInfo(AudioFrame* f, bool m) : frame(f), muted(m) {}
|
||||||
|
AudioFrame* frame;
|
||||||
|
bool muted;
|
||||||
|
};
|
||||||
|
|
||||||
|
typedef std::list<FrameAndMuteInfo> AudioFrameList;
|
||||||
|
typedef std::list<MixerAudioSource*> MixerAudioSourceList;
|
||||||
|
|
||||||
|
// Cheshire cat implementation of MixerAudioSource's non virtual functions.
|
||||||
|
class NewMixHistory {
|
||||||
|
public:
|
||||||
|
NewMixHistory();
|
||||||
|
~NewMixHistory();
|
||||||
|
|
||||||
|
// Returns true if the participant is being mixed.
|
||||||
|
bool IsMixed() const;
|
||||||
|
|
||||||
|
// Returns true if the participant was mixed previous mix
|
||||||
|
// iteration.
|
||||||
|
bool WasMixed() const;
|
||||||
|
|
||||||
|
// Updates the mixed status.
|
||||||
|
int32_t SetIsMixed(bool mixed);
|
||||||
|
|
||||||
|
void ResetMixedStatus();
|
||||||
|
|
||||||
|
private:
|
||||||
|
bool _isMixed;
|
||||||
|
};
|
||||||
|
|
||||||
|
class NewAudioConferenceMixerImpl : public NewAudioConferenceMixer {
|
||||||
|
public:
|
||||||
|
// AudioProcessing only accepts 10 ms frames.
|
||||||
|
enum { kProcessPeriodicityInMs = 10 };
|
||||||
|
|
||||||
|
explicit NewAudioConferenceMixerImpl(int id);
|
||||||
|
~NewAudioConferenceMixerImpl();
|
||||||
|
|
||||||
|
// Must be called after ctor.
|
||||||
|
bool Init();
|
||||||
|
|
||||||
|
// Module functions
|
||||||
|
int64_t TimeUntilNextProcess() override;
|
||||||
|
void Process() override;
|
||||||
|
|
||||||
|
// NewAudioConferenceMixer functions
|
||||||
|
int32_t RegisterMixedStreamCallback(
|
||||||
|
OldAudioMixerOutputReceiver* mixReceiver) override;
|
||||||
|
int32_t UnRegisterMixedStreamCallback() override;
|
||||||
|
int32_t SetMixabilityStatus(MixerAudioSource* participant,
|
||||||
|
bool mixable) override;
|
||||||
|
bool MixabilityStatus(const MixerAudioSource& participant) const override;
|
||||||
|
int32_t SetMinimumMixingFrequency(Frequency freq) override;
|
||||||
|
int32_t SetAnonymousMixabilityStatus(MixerAudioSource* participant,
|
||||||
|
bool mixable) override;
|
||||||
|
bool AnonymousMixabilityStatus(
|
||||||
|
const MixerAudioSource& participant) const override;
|
||||||
|
|
||||||
|
private:
|
||||||
|
enum { DEFAULT_AUDIO_FRAME_POOLSIZE = 50 };
|
||||||
|
|
||||||
|
// Set/get mix frequency
|
||||||
|
int32_t SetOutputFrequency(const Frequency& frequency);
|
||||||
|
Frequency OutputFrequency() const;
|
||||||
|
|
||||||
|
// Fills mixList with the AudioFrames pointers that should be used when
|
||||||
|
// mixing.
|
||||||
|
// maxAudioFrameCounter both input and output specifies how many more
|
||||||
|
// AudioFrames that are allowed to be mixed.
|
||||||
|
// rampOutList contain AudioFrames corresponding to an audio stream that
|
||||||
|
// used to be mixed but shouldn't be mixed any longer. These AudioFrames
|
||||||
|
// should be ramped out over this AudioFrame to avoid audio discontinuities.
|
||||||
|
void UpdateToMix(AudioFrameList* mixList,
|
||||||
|
AudioFrameList* rampOutList,
|
||||||
|
std::map<int, MixerAudioSource*>* mixParticipantList,
|
||||||
|
size_t* maxAudioFrameCounter) const;
|
||||||
|
|
||||||
|
// Return the lowest mixing frequency that can be used without having to
|
||||||
|
// downsample any audio.
|
||||||
|
int32_t GetLowestMixingFrequency() const;
|
||||||
|
int32_t GetLowestMixingFrequencyFromList(
|
||||||
|
const MixerAudioSourceList& mixList) const;
|
||||||
|
|
||||||
|
// Return the AudioFrames that should be mixed anonymously.
|
||||||
|
void GetAdditionalAudio(AudioFrameList* additionalFramesList) const;
|
||||||
|
|
||||||
|
// Update the NewMixHistory of all MixerAudioSources. mixedParticipantsList
|
||||||
|
// should contain a map of MixerAudioSources that have been mixed.
|
||||||
|
void UpdateMixedStatus(
|
||||||
|
const std::map<int, MixerAudioSource*>& mixedParticipantsList) const;
|
||||||
|
|
||||||
|
// Clears audioFrameList and reclaims all memory associated with it.
|
||||||
|
void ClearAudioFrameList(AudioFrameList* audioFrameList) const;
|
||||||
|
|
||||||
|
// This function returns true if it finds the MixerAudioSource in the
|
||||||
|
// specified list of MixerAudioSources.
|
||||||
|
bool IsParticipantInList(const MixerAudioSource& participant,
|
||||||
|
const MixerAudioSourceList& participantList) const;
|
||||||
|
|
||||||
|
// Add/remove the MixerAudioSource to the specified
|
||||||
|
// MixerAudioSource list.
|
||||||
|
bool AddParticipantToList(MixerAudioSource* participant,
|
||||||
|
MixerAudioSourceList* participantList) const;
|
||||||
|
bool RemoveParticipantFromList(MixerAudioSource* removeParticipant,
|
||||||
|
MixerAudioSourceList* participantList) const;
|
||||||
|
|
||||||
|
// Mix the AudioFrames stored in audioFrameList into mixedAudio.
|
||||||
|
int32_t MixFromList(AudioFrame* mixedAudio,
|
||||||
|
const AudioFrameList& audioFrameList) const;
|
||||||
|
|
||||||
|
// Mix the AudioFrames stored in audioFrameList into mixedAudio. No
|
||||||
|
// record will be kept of this mix (e.g. the corresponding MixerAudioSources
|
||||||
|
// will not be marked as IsMixed()
|
||||||
|
int32_t MixAnonomouslyFromList(AudioFrame* mixedAudio,
|
||||||
|
const AudioFrameList& audioFrameList) const;
|
||||||
|
|
||||||
|
bool LimitMixedAudio(AudioFrame* mixedAudio) const;
|
||||||
|
|
||||||
|
std::unique_ptr<CriticalSectionWrapper> _crit;
|
||||||
|
std::unique_ptr<CriticalSectionWrapper> _cbCrit;
|
||||||
|
|
||||||
|
int32_t _id;
|
||||||
|
|
||||||
|
Frequency _minimumMixingFreq;
|
||||||
|
|
||||||
|
// Mix result callback
|
||||||
|
OldAudioMixerOutputReceiver* _mixReceiver;
|
||||||
|
|
||||||
|
// The current sample frequency and sample size when mixing.
|
||||||
|
Frequency _outputFrequency;
|
||||||
|
size_t _sampleSize;
|
||||||
|
|
||||||
|
// Memory pool to avoid allocating/deallocating AudioFrames
|
||||||
|
MemoryPool<AudioFrame>* _audioFramePool;
|
||||||
|
|
||||||
|
// List of all participants. Note all lists are disjunct
|
||||||
|
MixerAudioSourceList _participantList; // May be mixed.
|
||||||
|
// Always mixed, anonomously.
|
||||||
|
MixerAudioSourceList _additionalParticipantList;
|
||||||
|
|
||||||
|
size_t _numMixedParticipants;
|
||||||
|
// Determines if we will use a limiter for clipping protection during
|
||||||
|
// mixing.
|
||||||
|
bool use_limiter_;
|
||||||
|
|
||||||
|
uint32_t _timeStamp;
|
||||||
|
|
||||||
|
// Metronome class.
|
||||||
|
TimeScheduler _timeScheduler;
|
||||||
|
|
||||||
|
// Counter keeping track of concurrent calls to process.
|
||||||
|
// Note: should never be higher than 1 or lower than 0.
|
||||||
|
int16_t _processCalls;
|
||||||
|
|
||||||
|
// Used for inhibiting saturation in mixing.
|
||||||
|
std::unique_ptr<AudioProcessing> _limiter;
|
||||||
|
};
|
||||||
|
} // namespace webrtc
|
||||||
|
|
||||||
|
#endif // WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
|
||||||
@ -12,6 +12,7 @@
|
|||||||
'audio_coding/audio_coding.gypi',
|
'audio_coding/audio_coding.gypi',
|
||||||
'audio_conference_mixer/audio_conference_mixer.gypi',
|
'audio_conference_mixer/audio_conference_mixer.gypi',
|
||||||
'audio_device/audio_device.gypi',
|
'audio_device/audio_device.gypi',
|
||||||
|
'audio_mixer/audio_mixer.gypi',
|
||||||
'audio_processing/audio_processing.gypi',
|
'audio_processing/audio_processing.gypi',
|
||||||
'bitrate_controller/bitrate_controller.gypi',
|
'bitrate_controller/bitrate_controller.gypi',
|
||||||
'congestion_controller/congestion_controller.gypi',
|
'congestion_controller/congestion_controller.gypi',
|
||||||
@ -126,6 +127,7 @@
|
|||||||
'audio_coding_module',
|
'audio_coding_module',
|
||||||
'audio_conference_mixer',
|
'audio_conference_mixer',
|
||||||
'audio_device' ,
|
'audio_device' ,
|
||||||
|
'audio_mixer',
|
||||||
'audio_processing',
|
'audio_processing',
|
||||||
'audioproc_test_utils',
|
'audioproc_test_utils',
|
||||||
'bitrate_controller',
|
'bitrate_controller',
|
||||||
|
|||||||
Reference in New Issue
Block a user