Remove voe::OutputMixer and AudioConferenceMixer.
This code path is not used anymore.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/3015553002
Cr-Commit-Position: refs/heads/master@{#19929}
diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index 88e4fb8..70c4759 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -22,7 +22,6 @@
'examples/objc',
'media',
'modules/audio_coding',
- 'modules/audio_conference_mixer',
'modules/audio_device',
'modules/audio_processing',
'modules/desktop_capture',
@@ -74,7 +73,6 @@
LEGACY_API_DIRS = (
'common_audio/include',
'modules/audio_coding/include',
- 'modules/audio_conference_mixer/include',
'modules/audio_processing/include',
'modules/bitrate_controller/include',
'modules/congestion_controller/include',
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 8ff1d58..37985d8 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -247,7 +247,7 @@
}
int AudioReceiveStream::PreferredSampleRate() const {
- return channel_proxy_->NeededFrequency();
+ return channel_proxy_->PreferredSampleRate();
}
int AudioReceiveStream::id() const {
diff --git a/modules/BUILD.gn b/modules/BUILD.gn
index 7394724..5d4987e 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -12,7 +12,6 @@
group("modules") {
public_deps = [
"audio_coding",
- "audio_conference_mixer",
"audio_device",
"audio_mixer",
"audio_processing",
@@ -232,7 +231,6 @@
":module_api",
"../test:test_main",
"audio_coding:audio_coding_unittests",
- "audio_conference_mixer:audio_conference_mixer_unittests",
"audio_device:audio_device_unittests",
"audio_mixer:audio_mixer_unittests",
"audio_processing:audio_processing_unittests",
diff --git a/modules/audio_conference_mixer/BUILD.gn b/modules/audio_conference_mixer/BUILD.gn
deleted file mode 100644
index 16a62b8..0000000
--- a/modules/audio_conference_mixer/BUILD.gn
+++ /dev/null
@@ -1,80 +0,0 @@
-# Copyright (c) 2014 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.
-
-import("../../webrtc.gni")
-
-config("audio_conference_mixer_config") {
- visibility = [ ":*" ] # Only targets in this file can depend on this.
- include_dirs = [
- "include",
- "../include",
- ]
-}
-
-rtc_static_library("audio_conference_mixer") {
- sources = [
- "include/audio_conference_mixer.h",
- "include/audio_conference_mixer_defines.h",
- "source/audio_conference_mixer_impl.cc",
- "source/audio_conference_mixer_impl.h",
- "source/audio_frame_manipulator.cc",
- "source/audio_frame_manipulator.h",
- "source/memory_pool.h",
- "source/memory_pool_posix.h",
- "source/memory_pool_win.h",
- "source/time_scheduler.cc",
- "source/time_scheduler.h",
- ]
-
- public_configs = [ ":audio_conference_mixer_config" ]
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-
- deps = [
- "..:module_api",
- "../..:webrtc_common",
- "../../audio/utility:audio_frame_operations",
- "../../rtc_base:rtc_base_approved",
- "../../system_wrappers",
- "../audio_processing",
- ]
-}
-
-if (rtc_include_tests) {
- rtc_source_set("audio_conference_mixer_unittests") {
- testonly = true
-
- # Skip restricting visibility on mobile platforms since the tests on those
- # gets additional generated targets which would require many lines here to
- # cover (which would be confusing to read and hard to maintain).
- if (!is_android && !is_ios) {
- visibility = [ "..:modules_unittests" ]
- }
- sources = [
- "test/audio_conference_mixer_unittest.cc",
- ]
- deps = [
- ":audio_conference_mixer",
- "../../test:test_support",
- "//testing/gmock",
- ]
- if (is_win) {
- cflags = [
- # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
- "/wd4373", # virtual function override.
- ]
- }
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
- }
-}
diff --git a/modules/audio_conference_mixer/DEPS b/modules/audio_conference_mixer/DEPS
deleted file mode 100644
index 2410757..0000000
--- a/modules/audio_conference_mixer/DEPS
+++ /dev/null
@@ -1,4 +0,0 @@
-include_rules = [
- "+audio/utility/audio_frame_operations.h",
- "+system_wrappers",
-]
diff --git a/modules/audio_conference_mixer/OWNERS b/modules/audio_conference_mixer/OWNERS
deleted file mode 100644
index 6df09bb..0000000
--- a/modules/audio_conference_mixer/OWNERS
+++ /dev/null
@@ -1,6 +0,0 @@
-minyue@webrtc.org
-
-# 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 *.gn=*
-per-file *.gni=*
diff --git a/modules/audio_conference_mixer/include/audio_conference_mixer.h b/modules/audio_conference_mixer/include/audio_conference_mixer.h
deleted file mode 100644
index 0ccb6d9..0000000
--- a/modules/audio_conference_mixer/include/audio_conference_mixer.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * 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 MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
-#define MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
-
-#include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
-#include "modules/include/module.h"
-#include "modules/include/module_common_types.h"
-
-namespace webrtc {
-class AudioMixerOutputReceiver;
-class MixerParticipant;
-class Trace;
-
-class AudioConferenceMixer : 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 AudioConferenceMixer* Create(int id);
- virtual ~AudioConferenceMixer() {}
-
- // 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(
- AudioMixerOutputReceiver* receiver) = 0;
- virtual int32_t UnRegisterMixedStreamCallback() = 0;
-
- // Add/remove participants as candidates for mixing.
- virtual int32_t SetMixabilityStatus(MixerParticipant* participant,
- bool mixable) = 0;
- // Returns true if a participant is a candidate for mixing.
- virtual bool MixabilityStatus(
- const MixerParticipant& 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(
- MixerParticipant* participant, bool mixable) = 0;
- // Returns true if the participant is mixed anonymously.
- virtual bool AnonymousMixabilityStatus(
- const MixerParticipant& 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:
- AudioConferenceMixer() {}
-};
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h b/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
deleted file mode 100644
index b40be23..0000000
--- a/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- * 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 MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-#define MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-
-#include "modules/include/module_common_types.h"
-#include "rtc_base/checks.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-class MixHistory;
-
-// A callback class that all mixer participants must inherit from/implement.
-class MixerParticipant
-{
-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;
-
- MixHistory* _mixHistory;
-protected:
- MixerParticipant();
- virtual ~MixerParticipant();
-};
-
-class AudioMixerOutputReceiver
-{
-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:
- AudioMixerOutputReceiver() {}
- virtual ~AudioMixerOutputReceiver() {}
-};
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
deleted file mode 100644
index 7927d24..0000000
--- a/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ /dev/null
@@ -1,904 +0,0 @@
-/*
- * 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 "modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
-#include "audio/utility/audio_frame_operations.h"
-#include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
-#include "modules/audio_conference_mixer/source/audio_frame_manipulator.h"
-#include "modules/audio_processing/include/audio_processing.h"
-#include "rtc_base/logging.h"
-
-namespace webrtc {
-namespace {
-
-struct ParticipantFrameStruct {
- ParticipantFrameStruct(MixerParticipant* p, AudioFrame* a, bool m)
- : participant(p), audioFrame(a), muted(m) {}
- MixerParticipant* 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) {
- // This is to avoid saturation in the mixing. It is only
- // meaningful if the limiter will be used.
- AudioFrameOperations::ApplyHalfGain(frame);
- }
- 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);
- }
-
- AudioFrameOperations::Add(*frame, mixed_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
-
-MixerParticipant::MixerParticipant()
- : _mixHistory(new MixHistory()) {
-}
-
-MixerParticipant::~MixerParticipant() {
- delete _mixHistory;
-}
-
-bool MixerParticipant::IsMixed() const {
- return _mixHistory->IsMixed();
-}
-
-MixHistory::MixHistory()
- : _isMixed(0) {
-}
-
-MixHistory::~MixHistory() {
-}
-
-bool MixHistory::IsMixed() const {
- return _isMixed;
-}
-
-bool MixHistory::WasMixed() const {
- // Was mixed is the same as is mixed depending on perspective. This function
- // is for the perspective of AudioConferenceMixerImpl.
- return IsMixed();
-}
-
-int32_t MixHistory::SetIsMixed(const bool mixed) {
- _isMixed = mixed;
- return 0;
-}
-
-void MixHistory::ResetMixedStatus() {
- _isMixed = false;
-}
-
-AudioConferenceMixer* AudioConferenceMixer::Create(int id) {
- AudioConferenceMixerImpl* mixer = new AudioConferenceMixerImpl(id);
- if(!mixer->Init()) {
- delete mixer;
- return NULL;
- }
- return mixer;
-}
-
-AudioConferenceMixerImpl::AudioConferenceMixerImpl(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 AudioConferenceMixerImpl::Init() {
- 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;
-}
-
-AudioConferenceMixerImpl::~AudioConferenceMixerImpl() {
- MemoryPool<AudioFrame>::DeleteMemoryPool(_audioFramePool);
- assert(_audioFramePool == NULL);
-}
-
-// Process should be called every kProcessPeriodicityInMs ms
-int64_t AudioConferenceMixerImpl::TimeUntilNextProcess() {
- int64_t timeUntilNextProcess = 0;
- rtc::CritScope cs(&_crit);
- if(_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0) {
- LOG(LS_ERROR) << "failed in TimeToNextUpdate() call";
- // Sanity check
- assert(false);
- return -1;
- }
- return timeUntilNextProcess;
-}
-
-void AudioConferenceMixerImpl::Process() {
- size_t remainingParticipantsAllowedToMix =
- kMaximumAmountOfMixedParticipants;
- {
- rtc::CritScope cs(&_crit);
- 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, MixerParticipant*> mixedParticipantsMap;
- {
- rtc::CritScope cs(&_cbCrit);
-
- 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) {
- rtc::CritScope cs(&_crit);
- _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);
-
- rtc::CritScope cs(&_crit);
- _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) {
- LOG(LS_ERROR) << "failed PopMemory() call";
- assert(false);
- return;
- }
-
- {
- rtc::CritScope cs(&_crit);
-
- // 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;
- AudioFrameOperations::Mute(mixedAudio);
- } else {
- // Only call the limiter if we have something to mix.
- LimitMixedAudio(mixedAudio);
- }
- }
-
- {
- rtc::CritScope cs(&_cbCrit);
- 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);
- {
- rtc::CritScope cs(&_crit);
- _processCalls--;
- }
- return;
-}
-
-int32_t AudioConferenceMixerImpl::RegisterMixedStreamCallback(
- AudioMixerOutputReceiver* mixReceiver) {
- rtc::CritScope cs(&_cbCrit);
- if(_mixReceiver != NULL) {
- return -1;
- }
- _mixReceiver = mixReceiver;
- return 0;
-}
-
-int32_t AudioConferenceMixerImpl::UnRegisterMixedStreamCallback() {
- rtc::CritScope cs(&_cbCrit);
- if(_mixReceiver == NULL) {
- return -1;
- }
- _mixReceiver = NULL;
- return 0;
-}
-
-int32_t AudioConferenceMixerImpl::SetOutputFrequency(
- const Frequency& frequency) {
- rtc::CritScope cs(&_crit);
-
- _outputFrequency = frequency;
- _sampleSize =
- static_cast<size_t>((_outputFrequency*kProcessPeriodicityInMs) / 1000);
-
- return 0;
-}
-
-AudioConferenceMixer::Frequency
-AudioConferenceMixerImpl::OutputFrequency() const {
- rtc::CritScope cs(&_crit);
- return _outputFrequency;
-}
-
-int32_t AudioConferenceMixerImpl::SetMixabilityStatus(
- MixerParticipant* 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;
- {
- rtc::CritScope cs(&_cbCrit);
- const bool isMixed =
- IsParticipantInList(*participant, _participantList);
- // API must be called with a new state.
- if(!(mixable ^ isMixed)) {
- LOG(LS_ERROR) << "Mixable is aready " <<
- (isMixed ? "ON" : "off");
- return -1;
- }
- bool success = false;
- if(mixable) {
- success = AddParticipantToList(participant, &_participantList);
- } else {
- success = RemoveParticipantFromList(participant, &_participantList);
- }
- if(!success) {
- LOG(LS_ERROR) << "failed to " << (mixable ? "add" : "remove")
- << " participant";
- assert(false);
- return -1;
- }
-
- size_t numMixedNonAnonymous = _participantList.size();
- if (numMixedNonAnonymous > kMaximumAmountOfMixedParticipants) {
- numMixedNonAnonymous = kMaximumAmountOfMixedParticipants;
- }
- numMixedParticipants =
- numMixedNonAnonymous + _additionalParticipantList.size();
- }
- // A MixerParticipant was added or removed. Make sure the scratch
- // buffer is updated if necessary.
- // Note: The scratch buffer may only be updated in Process().
- rtc::CritScope cs(&_crit);
- _numMixedParticipants = numMixedParticipants;
- return 0;
-}
-
-bool AudioConferenceMixerImpl::MixabilityStatus(
- const MixerParticipant& participant) const {
- rtc::CritScope cs(&_cbCrit);
- return IsParticipantInList(participant, _participantList);
-}
-
-int32_t AudioConferenceMixerImpl::SetAnonymousMixabilityStatus(
- MixerParticipant* participant, bool anonymous) {
- rtc::CritScope cs(&_cbCrit);
- if(IsParticipantInList(*participant, _additionalParticipantList)) {
- if(anonymous) {
- return 0;
- }
- if(!RemoveParticipantFromList(participant,
- &_additionalParticipantList)) {
- LOG(LS_ERROR) << "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) {
- LOG(LS_WARNING) <<
- "participant must be registered before turning it into anonymous";
- // Setting anonymous status is only possible if MixerParticipant is
- // already registered.
- return -1;
- }
- return AddParticipantToList(participant, &_additionalParticipantList) ?
- 0 : -1;
-}
-
-bool AudioConferenceMixerImpl::AnonymousMixabilityStatus(
- const MixerParticipant& participant) const {
- rtc::CritScope cs(&_cbCrit);
- return IsParticipantInList(participant, _additionalParticipantList);
-}
-
-int32_t AudioConferenceMixerImpl::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 {
- LOG(LS_ERROR) << "SetMinimumMixingFrequency incorrect frequency: "
- << 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 AudioConferenceMixerImpl::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 AudioConferenceMixerImpl::GetLowestMixingFrequencyFromList(
- const MixerParticipantList& mixList) const {
- int32_t highestFreq = 8000;
- for (MixerParticipantList::const_iterator iter = mixList.begin();
- iter != mixList.end();
- ++iter) {
- const int32_t neededFrequency = (*iter)->NeededFrequency(_id);
- if(neededFrequency > highestFreq) {
- highestFreq = neededFrequency;
- }
- }
- return highestFreq;
-}
-
-void AudioConferenceMixerImpl::UpdateToMix(
- AudioFrameList* mixList,
- AudioFrameList* rampOutList,
- std::map<int, MixerParticipant*>* mixParticipantList,
- size_t* maxAudioFrameCounter) const {
- LOG(LS_VERBOSE) <<
- "UpdateToMix(mixList,rampOutList,mixParticipantList," <<
- *maxAudioFrameCounter << ")";
- const size_t mixListStartSize = mixList->size();
- AudioFrameList activeList;
- // Struct needed by the passive lists to keep track of which AudioFrame
- // belongs to which MixerParticipant.
- ParticipantFrameStructList passiveWasNotMixedList;
- ParticipantFrameStructList passiveWasMixedList;
- for (MixerParticipantList::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) {
- LOG(LS_ERROR) << "failed PopMemory() call";
- assert(false);
- return;
- }
- audioFrame->sample_rate_hz_ = _outputFrequency;
-
- auto ret = (*participant)->GetAudioFrameWithMuted(_id, audioFrame);
- if (ret == MixerParticipant::AudioFrameInfo::kError) {
- LOG(LS_WARNING)
- << "failed to GetAudioFrameWithMuted() from participant";
- _audioFramePool->PushMemory(audioFrame);
- continue;
- }
- const bool muted = (ret == MixerParticipant::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) {
- LOG(LS_WARNING) << "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, MixerParticipant*>::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 AudioConferenceMixerImpl::GetAdditionalAudio(
- AudioFrameList* additionalFramesList) const {
- LOG(LS_VERBOSE) << "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.
- MixerParticipantList additionalParticipantList;
- additionalParticipantList.insert(additionalParticipantList.begin(),
- _additionalParticipantList.begin(),
- _additionalParticipantList.end());
-
- for (MixerParticipantList::const_iterator participant =
- additionalParticipantList.begin();
- participant != additionalParticipantList.end();
- ++participant) {
- AudioFrame* audioFrame = NULL;
- if(_audioFramePool->PopMemory(audioFrame) == -1) {
- LOG(LS_ERROR) << "failed PopMemory() call";
- assert(false);
- return;
- }
- audioFrame->sample_rate_hz_ = _outputFrequency;
- auto ret = (*participant)->GetAudioFrameWithMuted(_id, audioFrame);
- if (ret == MixerParticipant::AudioFrameInfo::kError) {
- LOG(LS_WARNING)
- << "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 == MixerParticipant::AudioFrameInfo::kMuted));
- }
-}
-
-void AudioConferenceMixerImpl::UpdateMixedStatus(
- const std::map<int, MixerParticipant*>& mixedParticipantsMap) const {
- LOG(LS_VERBOSE) << "UpdateMixedStatus(mixedParticipantsMap)";
- assert(mixedParticipantsMap.size() <= kMaximumAmountOfMixedParticipants);
-
- // Loop through all participants. If they are in the mix map they
- // were mixed.
- for (MixerParticipantList::const_iterator
- participant =_participantList.begin();
- participant != _participantList.end();
- ++participant) {
- bool isMixed = false;
- for (auto it = mixedParticipantsMap.begin();
- it != mixedParticipantsMap.end();
- ++it) {
- if (it->second == *participant) {
- isMixed = true;
- break;
- }
- }
- (*participant)->_mixHistory->SetIsMixed(isMixed);
- }
-}
-
-void AudioConferenceMixerImpl::ClearAudioFrameList(
- AudioFrameList* audioFrameList) const {
- LOG(LS_VERBOSE) << "ClearAudioFrameList(audioFrameList)";
- for (AudioFrameList::iterator iter = audioFrameList->begin();
- iter != audioFrameList->end();
- ++iter) {
- _audioFramePool->PushMemory(iter->frame);
- }
- audioFrameList->clear();
-}
-
-bool AudioConferenceMixerImpl::IsParticipantInList(
- const MixerParticipant& participant,
- const MixerParticipantList& participantList) const {
- LOG(LS_VERBOSE) << "IsParticipantInList(participant,participantList)";
- for (MixerParticipantList::const_iterator iter = participantList.begin();
- iter != participantList.end();
- ++iter) {
- if(&participant == *iter) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioConferenceMixerImpl::AddParticipantToList(
- MixerParticipant* participant,
- MixerParticipantList* participantList) const {
- LOG(LS_VERBOSE) << "AddParticipantToList(participant, participantList)";
- participantList->push_back(participant);
- // Make sure that the mixed status is correct for new MixerParticipant.
- participant->_mixHistory->ResetMixedStatus();
- return true;
-}
-
-bool AudioConferenceMixerImpl::RemoveParticipantFromList(
- MixerParticipant* participant,
- MixerParticipantList* participantList) const {
- LOG(LS_VERBOSE)
- << "RemoveParticipantFromList(participant, participantList)";
- for (MixerParticipantList::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 AudioConferenceMixerImpl::MixFromList(
- AudioFrame* mixedAudio,
- const AudioFrameList& audioFrameList) const {
-
- LOG(LS_VERBOSE) << "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) {
- LOG(LS_ERROR) <<
- "Trying to mix more than max amount of mixed participants:"
- << 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 AudioConferenceMixerImpl::MixAnonomouslyFromList(
- AudioFrame* mixedAudio,
- const AudioFrameList& audioFrameList) const {
- LOG(LS_VERBOSE) << "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 AudioConferenceMixerImpl::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).
- AudioFrameOperations::Add(*mixedAudio, mixedAudio);
-
- if(error != _limiter->kNoError) {
- LOG(LS_ERROR) << "Error from AudioProcessing: " << error;
- assert(false);
- return false;
- }
- return true;
-}
-} // namespace webrtc
diff --git a/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
deleted file mode 100644
index e88cb19..0000000
--- a/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
+++ /dev/null
@@ -1,192 +0,0 @@
-/*
- * 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 MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
-#define MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
-
-#include <list>
-#include <map>
-#include <memory>
-
-#include "modules/audio_conference_mixer/include/audio_conference_mixer.h"
-#include "modules/audio_conference_mixer/source/memory_pool.h"
-#include "modules/audio_conference_mixer/source/time_scheduler.h"
-#include "modules/include/module_common_types.h"
-#include "rtc_base/criticalsection.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-class AudioProcessing;
-
-struct FrameAndMuteInfo {
- FrameAndMuteInfo(AudioFrame* f, bool m) : frame(f), muted(m) {}
- AudioFrame* frame;
- bool muted;
-};
-
-typedef std::list<FrameAndMuteInfo> AudioFrameList;
-typedef std::list<MixerParticipant*> MixerParticipantList;
-
-// Cheshire cat implementation of MixerParticipant's non virtual functions.
-class MixHistory
-{
-public:
- MixHistory();
- ~MixHistory();
-
- // 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 AudioConferenceMixerImpl : public AudioConferenceMixer
-{
-public:
- // AudioProcessing only accepts 10 ms frames.
- enum {kProcessPeriodicityInMs = 10};
-
- AudioConferenceMixerImpl(int id);
- ~AudioConferenceMixerImpl();
-
- // Must be called after ctor.
- bool Init();
-
- // Module functions
- int64_t TimeUntilNextProcess() override;
- void Process() override;
-
- // AudioConferenceMixer functions
- int32_t RegisterMixedStreamCallback(
- AudioMixerOutputReceiver* mixReceiver) override;
- int32_t UnRegisterMixedStreamCallback() override;
- int32_t SetMixabilityStatus(MixerParticipant* participant,
- bool mixable) override;
- bool MixabilityStatus(const MixerParticipant& participant) const override;
- int32_t SetMinimumMixingFrequency(Frequency freq) override;
- int32_t SetAnonymousMixabilityStatus(
- MixerParticipant* participant, bool mixable) override;
- bool AnonymousMixabilityStatus(
- const MixerParticipant& 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, MixerParticipant*>* 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 MixerParticipantList& mixList) const;
-
- // Return the AudioFrames that should be mixed anonymously.
- void GetAdditionalAudio(AudioFrameList* additionalFramesList) const;
-
- // Update the MixHistory of all MixerParticipants. mixedParticipantsList
- // should contain a map of MixerParticipants that have been mixed.
- void UpdateMixedStatus(
- const std::map<int, MixerParticipant*>& mixedParticipantsList) const;
-
- // Clears audioFrameList and reclaims all memory associated with it.
- void ClearAudioFrameList(AudioFrameList* audioFrameList) const;
-
- // This function returns true if it finds the MixerParticipant in the
- // specified list of MixerParticipants.
- bool IsParticipantInList(const MixerParticipant& participant,
- const MixerParticipantList& participantList) const;
-
- // Add/remove the MixerParticipant to the specified
- // MixerParticipant list.
- bool AddParticipantToList(
- MixerParticipant* participant,
- MixerParticipantList* participantList) const;
- bool RemoveParticipantFromList(
- MixerParticipant* removeParticipant,
- MixerParticipantList* 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 MixerParticipants
- // will not be marked as IsMixed()
- int32_t MixAnonomouslyFromList(AudioFrame* mixedAudio,
- const AudioFrameList& audioFrameList) const;
-
- bool LimitMixedAudio(AudioFrame* mixedAudio) const;
-
- rtc::CriticalSection _crit;
- rtc::CriticalSection _cbCrit;
-
- int32_t _id;
-
- Frequency _minimumMixingFreq;
-
- // Mix result callback
- AudioMixerOutputReceiver* _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
- MixerParticipantList _participantList; // May be mixed.
- // Always mixed, anonomously.
- MixerParticipantList _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 // MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
diff --git a/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
deleted file mode 100644
index a16afb7..0000000
--- a/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- * 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 "modules/audio_conference_mixer/source/audio_frame_manipulator.h"
-#include "modules/include/module_common_types.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace {
-// Linear ramping over 80 samples.
-// TODO(hellner): ramp using fix point?
-const float rampArray[] = {0.0000f, 0.0127f, 0.0253f, 0.0380f,
- 0.0506f, 0.0633f, 0.0759f, 0.0886f,
- 0.1013f, 0.1139f, 0.1266f, 0.1392f,
- 0.1519f, 0.1646f, 0.1772f, 0.1899f,
- 0.2025f, 0.2152f, 0.2278f, 0.2405f,
- 0.2532f, 0.2658f, 0.2785f, 0.2911f,
- 0.3038f, 0.3165f, 0.3291f, 0.3418f,
- 0.3544f, 0.3671f, 0.3797f, 0.3924f,
- 0.4051f, 0.4177f, 0.4304f, 0.4430f,
- 0.4557f, 0.4684f, 0.4810f, 0.4937f,
- 0.5063f, 0.5190f, 0.5316f, 0.5443f,
- 0.5570f, 0.5696f, 0.5823f, 0.5949f,
- 0.6076f, 0.6203f, 0.6329f, 0.6456f,
- 0.6582f, 0.6709f, 0.6835f, 0.6962f,
- 0.7089f, 0.7215f, 0.7342f, 0.7468f,
- 0.7595f, 0.7722f, 0.7848f, 0.7975f,
- 0.8101f, 0.8228f, 0.8354f, 0.8481f,
- 0.8608f, 0.8734f, 0.8861f, 0.8987f,
- 0.9114f, 0.9241f, 0.9367f, 0.9494f,
- 0.9620f, 0.9747f, 0.9873f, 1.0000f};
-const size_t rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
-} // namespace
-
-namespace webrtc {
-uint32_t CalculateEnergy(const AudioFrame& audioFrame)
-{
- if (audioFrame.muted()) return 0;
-
- uint32_t energy = 0;
- const int16_t* frame_data = audioFrame.data();
- for(size_t position = 0; position < audioFrame.samples_per_channel_;
- position++)
- {
- // TODO(andrew): this can easily overflow.
- energy += frame_data[position] * frame_data[position];
- }
- return energy;
-}
-
-void RampIn(AudioFrame& audioFrame)
-{
- assert(rampSize <= audioFrame.samples_per_channel_);
- if (audioFrame.muted()) return;
-
- int16_t* frame_data = audioFrame.mutable_data();
- for(size_t i = 0; i < rampSize; i++)
- {
- frame_data[i] = static_cast<int16_t>(rampArray[i] * frame_data[i]);
- }
-}
-
-void RampOut(AudioFrame& audioFrame)
-{
- assert(rampSize <= audioFrame.samples_per_channel_);
- if (audioFrame.muted()) return;
-
- int16_t* frame_data = audioFrame.mutable_data();
- for(size_t i = 0; i < rampSize; i++)
- {
- const size_t rampPos = rampSize - 1 - i;
- frame_data[i] = static_cast<int16_t>(rampArray[rampPos] *
- frame_data[i]);
- }
- memset(&frame_data[rampSize], 0,
- (audioFrame.samples_per_channel_ - rampSize) *
- sizeof(frame_data[0]));
-}
-} // namespace webrtc
diff --git a/modules/audio_conference_mixer/source/audio_frame_manipulator.h b/modules/audio_conference_mixer/source/audio_frame_manipulator.h
deleted file mode 100644
index f8fe2d8..0000000
--- a/modules/audio_conference_mixer/source/audio_frame_manipulator.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * 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 MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_
-#define MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_
-
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-class AudioFrame;
-
-// Updates the audioFrame's energy (based on its samples).
-uint32_t CalculateEnergy(const AudioFrame& audioFrame);
-
-// Apply linear step function that ramps in/out the audio samples in audioFrame
-void RampIn(AudioFrame& audioFrame);
-void RampOut(AudioFrame& audioFrame);
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_
diff --git a/modules/audio_conference_mixer/source/memory_pool.h b/modules/audio_conference_mixer/source/memory_pool.h
deleted file mode 100644
index 4e93596..0000000
--- a/modules/audio_conference_mixer/source/memory_pool.h
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * 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 MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_
-#define MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_
-
-#include <assert.h>
-
-#include "typedefs.h" // NOLINT(build/include)
-
-#ifdef _WIN32
-#include "modules/audio_conference_mixer/source/memory_pool_win.h"
-#else
-#include "modules/audio_conference_mixer/source/memory_pool_posix.h"
-#endif
-
-namespace webrtc {
-
-template<class MemoryType>
-class MemoryPool
-{
-public:
- // Factory method, constructor disabled.
- static int32_t CreateMemoryPool(MemoryPool*& memoryPool,
- uint32_t initialPoolSize);
-
- // Try to delete the memory pool. Fail with return value -1 if there is
- // outstanding memory.
- static int32_t DeleteMemoryPool(
- MemoryPool*& memoryPool);
-
- // Get/return unused memory.
- int32_t PopMemory(MemoryType*& memory);
- int32_t PushMemory(MemoryType*& memory);
-private:
- MemoryPool(int32_t initialPoolSize);
- ~MemoryPool();
-
- MemoryPoolImpl<MemoryType>* _ptrImpl;
-};
-
-template<class MemoryType>
-MemoryPool<MemoryType>::MemoryPool(int32_t initialPoolSize)
-{
- _ptrImpl = new MemoryPoolImpl<MemoryType>(initialPoolSize);
-}
-
-template<class MemoryType>
-MemoryPool<MemoryType>::~MemoryPool()
-{
- delete _ptrImpl;
-}
-
-template<class MemoryType> int32_t
-MemoryPool<MemoryType>::CreateMemoryPool(MemoryPool*& memoryPool,
- uint32_t initialPoolSize)
-{
- memoryPool = new MemoryPool(initialPoolSize);
- if(memoryPool == NULL)
- {
- return -1;
- }
- if(memoryPool->_ptrImpl == NULL)
- {
- delete memoryPool;
- memoryPool = NULL;
- return -1;
- }
- if(!memoryPool->_ptrImpl->Initialize())
- {
- delete memoryPool;
- memoryPool = NULL;
- return -1;
- }
- return 0;
-}
-
-template<class MemoryType>
-int32_t MemoryPool<MemoryType>::DeleteMemoryPool(MemoryPool*& memoryPool)
-{
- if(memoryPool == NULL)
- {
- return -1;
- }
- if(memoryPool->_ptrImpl == NULL)
- {
- return -1;
- }
- if(memoryPool->_ptrImpl->Terminate() == -1)
- {
- return -1;
- }
- delete memoryPool;
- memoryPool = NULL;
- return 0;
-}
-
-template<class MemoryType>
-int32_t MemoryPool<MemoryType>::PopMemory(MemoryType*& memory)
-{
- return _ptrImpl->PopMemory(memory);
-}
-
-template<class MemoryType>
-int32_t MemoryPool<MemoryType>::PushMemory(MemoryType*& memory)
-{
- if(memory == NULL)
- {
- return -1;
- }
- return _ptrImpl->PushMemory(memory);
-}
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_
diff --git a/modules/audio_conference_mixer/source/memory_pool_posix.h b/modules/audio_conference_mixer/source/memory_pool_posix.h
deleted file mode 100644
index 70e2bfd..0000000
--- a/modules/audio_conference_mixer/source/memory_pool_posix.h
+++ /dev/null
@@ -1,156 +0,0 @@
-/*
- * 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 MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_
-#define MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_
-
-#include <assert.h>
-#include <list>
-
-#include "rtc_base/criticalsection.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-template<class MemoryType>
-class MemoryPoolImpl
-{
-public:
- // MemoryPool functions.
- int32_t PopMemory(MemoryType*& memory);
- int32_t PushMemory(MemoryType*& memory);
-
- MemoryPoolImpl(int32_t initialPoolSize);
- ~MemoryPoolImpl();
-
- // Atomic functions
- int32_t Terminate();
- bool Initialize();
-private:
- // Non-atomic function.
- int32_t CreateMemory(uint32_t amountToCreate);
-
- rtc::CriticalSection _crit;
-
- bool _terminate;
-
- std::list<MemoryType*> _memoryPool;
-
- uint32_t _initialPoolSize;
- uint32_t _createdMemory;
- uint32_t _outstandingMemory;
-};
-
-template<class MemoryType>
-MemoryPoolImpl<MemoryType>::MemoryPoolImpl(int32_t initialPoolSize)
- : _terminate(false),
- _initialPoolSize(initialPoolSize),
- _createdMemory(0),
- _outstandingMemory(0)
-{
-}
-
-template<class MemoryType>
-MemoryPoolImpl<MemoryType>::~MemoryPoolImpl()
-{
- // Trigger assert if there is outstanding memory.
- assert(_createdMemory == 0);
- assert(_outstandingMemory == 0);
-}
-
-template<class MemoryType>
-int32_t MemoryPoolImpl<MemoryType>::PopMemory(MemoryType*& memory)
-{
- rtc::CritScope cs(&_crit);
- if(_terminate)
- {
- memory = NULL;
- return -1;
- }
- if (_memoryPool.empty()) {
- // _memoryPool empty create new memory.
- CreateMemory(_initialPoolSize);
- if(_memoryPool.empty())
- {
- memory = NULL;
- return -1;
- }
- }
- memory = _memoryPool.front();
- _memoryPool.pop_front();
- _outstandingMemory++;
- return 0;
-}
-
-template<class MemoryType>
-int32_t MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory)
-{
- if(memory == NULL)
- {
- return -1;
- }
- rtc::CritScope cs(&_crit);
- _outstandingMemory--;
- if(_memoryPool.size() > (_initialPoolSize << 1))
- {
- // Reclaim memory if less than half of the pool is unused.
- _createdMemory--;
- delete memory;
- memory = NULL;
- return 0;
- }
- _memoryPool.push_back(memory);
- memory = NULL;
- return 0;
-}
-
-template<class MemoryType>
-bool MemoryPoolImpl<MemoryType>::Initialize()
-{
- rtc::CritScope cs(&_crit);
- return CreateMemory(_initialPoolSize) == 0;
-}
-
-template<class MemoryType>
-int32_t MemoryPoolImpl<MemoryType>::Terminate()
-{
- rtc::CritScope cs(&_crit);
- assert(_createdMemory == _outstandingMemory + _memoryPool.size());
-
- _terminate = true;
- // Reclaim all memory.
- while(_createdMemory > 0)
- {
- MemoryType* memory = _memoryPool.front();
- _memoryPool.pop_front();
- delete memory;
- _createdMemory--;
- }
- return 0;
-}
-
-template<class MemoryType>
-int32_t MemoryPoolImpl<MemoryType>::CreateMemory(
- uint32_t amountToCreate)
-{
- for(uint32_t i = 0; i < amountToCreate; i++)
- {
- MemoryType* memory = new MemoryType();
- if(memory == NULL)
- {
- return -1;
- }
- _memoryPool.push_back(memory);
- _createdMemory++;
- }
- return 0;
-}
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_
diff --git a/modules/audio_conference_mixer/source/memory_pool_win.h b/modules/audio_conference_mixer/source/memory_pool_win.h
deleted file mode 100644
index cce38aa..0000000
--- a/modules/audio_conference_mixer/source/memory_pool_win.h
+++ /dev/null
@@ -1,199 +0,0 @@
-/*
- * 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 MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_
-#define MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_
-
-#include <assert.h>
-#include <windows.h>
-
-#include "system_wrappers/include/aligned_malloc.h"
-#include "system_wrappers/include/atomic32.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-template<class MemoryType> struct MemoryPoolItem;
-
-template<class MemoryType>
-struct MemoryPoolItemPayload
-{
- MemoryPoolItemPayload()
- : memoryType(),
- base(NULL)
- {
- }
- MemoryType memoryType;
- MemoryPoolItem<MemoryType>* base;
-};
-
-template<class MemoryType>
-struct MemoryPoolItem
-{
- // Atomic single linked list entry header.
- SLIST_ENTRY itemEntry;
- // Atomic single linked list payload.
- MemoryPoolItemPayload<MemoryType>* payload;
-};
-
-template<class MemoryType>
-class MemoryPoolImpl
-{
-public:
- // MemoryPool functions.
- int32_t PopMemory(MemoryType*& memory);
- int32_t PushMemory(MemoryType*& memory);
-
- MemoryPoolImpl(int32_t /*initialPoolSize*/);
- ~MemoryPoolImpl();
-
- // Atomic functions.
- int32_t Terminate();
- bool Initialize();
-private:
- // Non-atomic function.
- MemoryPoolItem<MemoryType>* CreateMemory();
-
- // Windows implementation of single linked atomic list, documented here:
- // http://msdn.microsoft.com/en-us/library/ms686962(VS.85).aspx
-
- // Atomic single linked list head.
- PSLIST_HEADER _pListHead;
-
- Atomic32 _createdMemory;
- Atomic32 _outstandingMemory;
-};
-
-template<class MemoryType>
-MemoryPoolImpl<MemoryType>::MemoryPoolImpl(
- int32_t /*initialPoolSize*/)
- : _pListHead(NULL),
- _createdMemory(0),
- _outstandingMemory(0)
-{
-}
-
-template<class MemoryType>
-MemoryPoolImpl<MemoryType>::~MemoryPoolImpl()
-{
- Terminate();
- if(_pListHead != NULL)
- {
- AlignedFree(reinterpret_cast<void*>(_pListHead));
- _pListHead = NULL;
- }
- // Trigger assert if there is outstanding memory.
- assert(_createdMemory.Value() == 0);
- assert(_outstandingMemory.Value() == 0);
-}
-
-template<class MemoryType>
-int32_t MemoryPoolImpl<MemoryType>::PopMemory(MemoryType*& memory)
-{
- PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
- if(pListEntry == NULL)
- {
- MemoryPoolItem<MemoryType>* item = CreateMemory();
- if(item == NULL)
- {
- return -1;
- }
- pListEntry = &(item->itemEntry);
- }
- ++_outstandingMemory;
- memory = &((MemoryPoolItem<MemoryType>*)pListEntry)->payload->memoryType;
- return 0;
-}
-
-template<class MemoryType>
-int32_t MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory)
-{
- if(memory == NULL)
- {
- return -1;
- }
-
- MemoryPoolItem<MemoryType>* item =
- ((MemoryPoolItemPayload<MemoryType>*)memory)->base;
-
- const int32_t usedItems = --_outstandingMemory;
- const int32_t totalItems = _createdMemory.Value();
- const int32_t freeItems = totalItems - usedItems;
- if(freeItems < 0)
- {
- assert(false);
- delete item->payload;
- AlignedFree(item);
- return -1;
- }
- if(freeItems >= totalItems>>1)
- {
- delete item->payload;
- AlignedFree(item);
- --_createdMemory;
- return 0;
- }
- InterlockedPushEntrySList(_pListHead,&(item->itemEntry));
- return 0;
-}
-
-template<class MemoryType>
-bool MemoryPoolImpl<MemoryType>::Initialize()
-{
- _pListHead = (PSLIST_HEADER)AlignedMalloc(sizeof(SLIST_HEADER),
- MEMORY_ALLOCATION_ALIGNMENT);
- if(_pListHead == NULL)
- {
- return false;
- }
- InitializeSListHead(_pListHead);
- return true;
-}
-
-template<class MemoryType>
-int32_t MemoryPoolImpl<MemoryType>::Terminate()
-{
- int32_t itemsFreed = 0;
- PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
- while(pListEntry != NULL)
- {
- MemoryPoolItem<MemoryType>* item = ((MemoryPoolItem<MemoryType>*)pListEntry);
- delete item->payload;
- AlignedFree(item);
- --_createdMemory;
- itemsFreed++;
- pListEntry = InterlockedPopEntrySList(_pListHead);
- }
- return itemsFreed;
-}
-
-template<class MemoryType>
-MemoryPoolItem<MemoryType>* MemoryPoolImpl<MemoryType>::CreateMemory()
-{
- MemoryPoolItem<MemoryType>* returnValue = (MemoryPoolItem<MemoryType>*)
- AlignedMalloc(sizeof(MemoryPoolItem<MemoryType>),
- MEMORY_ALLOCATION_ALIGNMENT);
- if(returnValue == NULL)
- {
- return NULL;
- }
-
- returnValue->payload = new MemoryPoolItemPayload<MemoryType>();
- if(returnValue->payload == NULL)
- {
- delete returnValue;
- return NULL;
- }
- returnValue->payload->base = returnValue;
- ++_createdMemory;
- return returnValue;
-}
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_
diff --git a/modules/audio_conference_mixer/source/time_scheduler.cc b/modules/audio_conference_mixer/source/time_scheduler.cc
deleted file mode 100644
index 5a27b42..0000000
--- a/modules/audio_conference_mixer/source/time_scheduler.cc
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- * 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.
- */
-
-#include "modules/audio_conference_mixer/source/time_scheduler.h"
-#include "rtc_base/timeutils.h"
-
-namespace webrtc {
-TimeScheduler::TimeScheduler(const int64_t periodicityInMs)
- : _isStarted(false),
- _lastPeriodMark(),
- _periodicityInMs(periodicityInMs),
- _periodicityInTicks(periodicityInMs * rtc::kNumNanosecsPerMillisec),
- _missedPeriods(0) {}
-
-int32_t TimeScheduler::UpdateScheduler() {
- rtc::CritScope cs(&_crit);
- if(!_isStarted)
- {
- _isStarted = true;
- _lastPeriodMark = rtc::TimeNanos();
- return 0;
- }
- // Don't perform any calculations until the debt of pending periods have
- // been worked off.
- if(_missedPeriods > 0)
- {
- _missedPeriods--;
- return 0;
- }
-
- // Calculate the time that has past since previous call to this function.
- int64_t tickNow = rtc::TimeNanos();
- int64_t amassedTicks = tickNow - _lastPeriodMark;
- int64_t amassedMs = amassedTicks / rtc::kNumNanosecsPerMillisec;
-
- // Calculate the number of periods the time that has passed correspond to.
- int64_t periodsToClaim = amassedMs / _periodicityInMs;
-
- // One period will be worked off by this call. Make sure that the number of
- // pending periods don't end up being negative (e.g. if this function is
- // called to often).
- if(periodsToClaim < 1)
- {
- periodsToClaim = 1;
- }
-
- // Update the last period mark without introducing any drifting.
- // Note that if this fuunction is called to often _lastPeriodMark can
- // refer to a time in the future which in turn will yield TimeToNextUpdate
- // that is greater than the periodicity
- for(int64_t i = 0; i < periodsToClaim; i++)
- {
- _lastPeriodMark += _periodicityInTicks;
- }
-
- // Update the total amount of missed periods note that we have processed
- // one period hence the - 1
- _missedPeriods += periodsToClaim - 1;
- return 0;
-}
-
-int32_t TimeScheduler::TimeToNextUpdate(
- int64_t& updateTimeInMS) const
-{
- rtc::CritScope cs(&_crit);
- // Missed periods means that the next UpdateScheduler() should happen
- // immediately.
- if(_missedPeriods > 0)
- {
- updateTimeInMS = 0;
- return 0;
- }
-
- // Calculate the time (in ms) that has past since last call to
- // UpdateScheduler()
- int64_t tickNow = rtc::TimeNanos();
- int64_t ticksSinceLastUpdate = tickNow - _lastPeriodMark;
- const int64_t millisecondsSinceLastUpdate =
- ticksSinceLastUpdate / rtc::kNumNanosecsPerMillisec;
-
- updateTimeInMS = _periodicityInMs - millisecondsSinceLastUpdate;
- updateTimeInMS = (updateTimeInMS < 0) ? 0 : updateTimeInMS;
- return 0;
-}
-} // namespace webrtc
diff --git a/modules/audio_conference_mixer/source/time_scheduler.h b/modules/audio_conference_mixer/source/time_scheduler.h
deleted file mode 100644
index 1580861..0000000
--- a/modules/audio_conference_mixer/source/time_scheduler.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * 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.
- */
-
-// The TimeScheduler class keeps track of periodic events. It is non-drifting
-// and keeps track of any missed periods so that it is possible to catch up.
-// (compare to a metronome)
-#include "rtc_base/criticalsection.h"
-
-#ifndef MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
-#define MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
-
-namespace webrtc {
-
-class TimeScheduler {
-public:
- TimeScheduler(const int64_t periodicityInMs);
- ~TimeScheduler() = default;
-
- // Signal that a periodic event has been triggered.
- int32_t UpdateScheduler();
-
- // Set updateTimeInMs to the amount of time until UpdateScheduler() should
- // be called. This time will never be negative.
- int32_t TimeToNextUpdate(int64_t& updateTimeInMS) const;
-
-private:
- rtc::CriticalSection _crit;
-
- bool _isStarted;
- int64_t _lastPeriodMark; // In ns
-
- int64_t _periodicityInMs;
- int64_t _periodicityInTicks;
- uint32_t _missedPeriods;
-};
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
diff --git a/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc b/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
deleted file mode 100644
index 6685899..0000000
--- a/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- * Copyright (c) 2015 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 <memory>
-
-#include "modules/audio_conference_mixer/include/audio_conference_mixer.h"
-#include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
-#include "test/gmock.h"
-
-namespace webrtc {
-
-using testing::_;
-using testing::AtLeast;
-using testing::Invoke;
-using testing::Return;
-
-class MockAudioMixerOutputReceiver : public AudioMixerOutputReceiver {
- public:
- MOCK_METHOD4(NewMixedAudio, void(const int32_t id,
- const AudioFrame& general_audio_frame,
- const AudioFrame** unique_audio_frames,
- const uint32_t size));
-};
-
-class MockMixerParticipant : public MixerParticipant {
- public:
- MockMixerParticipant() {
- ON_CALL(*this, GetAudioFrame(_, _))
- .WillByDefault(Invoke(this, &MockMixerParticipant::FakeAudioFrame));
- }
- MOCK_METHOD2(GetAudioFrame,
- int32_t(const int32_t id, AudioFrame* audio_frame));
- MOCK_CONST_METHOD1(NeededFrequency, int32_t(const int32_t id));
- AudioFrame* fake_frame() { return &fake_frame_; }
-
- private:
- AudioFrame fake_frame_;
- int32_t FakeAudioFrame(const int32_t id, AudioFrame* audio_frame) {
- audio_frame->CopyFrom(fake_frame_);
- return 0;
- }
-};
-
-TEST(AudioConferenceMixer, AnonymousAndNamed) {
- const int kId = 1;
- // Should not matter even if partipants are more than
- // kMaximumAmountOfMixedParticipants.
- const int kNamed =
- AudioConferenceMixer::kMaximumAmountOfMixedParticipants + 1;
- const int kAnonymous =
- AudioConferenceMixer::kMaximumAmountOfMixedParticipants + 1;
-
- std::unique_ptr<AudioConferenceMixer> mixer(
- AudioConferenceMixer::Create(kId));
-
- MockMixerParticipant named[kNamed];
- MockMixerParticipant anonymous[kAnonymous];
-
- for (int i = 0; i < kNamed; ++i) {
- EXPECT_EQ(0, mixer->SetMixabilityStatus(&named[i], true));
- EXPECT_TRUE(mixer->MixabilityStatus(named[i]));
- }
-
- for (int i = 0; i < kAnonymous; ++i) {
- // Participant must be registered before turning it into anonymous.
- EXPECT_EQ(-1, mixer->SetAnonymousMixabilityStatus(&anonymous[i], true));
- EXPECT_EQ(0, mixer->SetMixabilityStatus(&anonymous[i], true));
- EXPECT_TRUE(mixer->MixabilityStatus(anonymous[i]));
- EXPECT_FALSE(mixer->AnonymousMixabilityStatus(anonymous[i]));
-
- EXPECT_EQ(0, mixer->SetAnonymousMixabilityStatus(&anonymous[i], true));
- EXPECT_TRUE(mixer->AnonymousMixabilityStatus(anonymous[i]));
-
- // Anonymous participants do not show status by MixabilityStatus.
- EXPECT_FALSE(mixer->MixabilityStatus(anonymous[i]));
- }
-
- for (int i = 0; i < kNamed; ++i) {
- EXPECT_EQ(0, mixer->SetMixabilityStatus(&named[i], false));
- EXPECT_FALSE(mixer->MixabilityStatus(named[i]));
- }
-
- for (int i = 0; i < kAnonymous - 1; i++) {
- EXPECT_EQ(0, mixer->SetAnonymousMixabilityStatus(&anonymous[i], false));
- EXPECT_FALSE(mixer->AnonymousMixabilityStatus(anonymous[i]));
-
- // SetAnonymousMixabilityStatus(anonymous, false) moves anonymous to the
- // named group.
- EXPECT_TRUE(mixer->MixabilityStatus(anonymous[i]));
- }
-
- // SetMixabilityStatus(anonymous, false) will remove anonymous from both
- // anonymous and named groups.
- EXPECT_EQ(0, mixer->SetMixabilityStatus(&anonymous[kAnonymous - 1], false));
- EXPECT_FALSE(mixer->AnonymousMixabilityStatus(anonymous[kAnonymous - 1]));
- EXPECT_FALSE(mixer->MixabilityStatus(anonymous[kAnonymous - 1]));
-}
-
-TEST(AudioConferenceMixer, LargestEnergyVadActiveMixed) {
- const int kId = 1;
- const int kParticipants =
- AudioConferenceMixer::kMaximumAmountOfMixedParticipants + 3;
- const int kSampleRateHz = 32000;
-
- std::unique_ptr<AudioConferenceMixer> mixer(
- AudioConferenceMixer::Create(kId));
-
- MockAudioMixerOutputReceiver output_receiver;
- EXPECT_EQ(0, mixer->RegisterMixedStreamCallback(&output_receiver));
-
- MockMixerParticipant participants[kParticipants];
-
- for (int i = 0; i < kParticipants; ++i) {
- participants[i].fake_frame()->id_ = i;
- participants[i].fake_frame()->sample_rate_hz_ = kSampleRateHz;
- participants[i].fake_frame()->speech_type_ = AudioFrame::kNormalSpeech;
- participants[i].fake_frame()->vad_activity_ = AudioFrame::kVadActive;
- participants[i].fake_frame()->num_channels_ = 1;
-
- // Frame duration 10ms.
- participants[i].fake_frame()->samples_per_channel_ = kSampleRateHz / 100;
-
- // We set the 80-th sample value since the first 80 samples may be
- // modified by a ramped-in window.
- participants[i].fake_frame()->mutable_data()[80] = i;
-
- EXPECT_EQ(0, mixer->SetMixabilityStatus(&participants[i], true));
- EXPECT_CALL(participants[i], GetAudioFrame(_, _))
- .Times(AtLeast(1));
- EXPECT_CALL(participants[i], NeededFrequency(_))
- .WillRepeatedly(Return(kSampleRateHz));
- }
-
- // Last participant gives audio frame with passive VAD, although it has the
- // largest energy.
- participants[kParticipants - 1].fake_frame()->vad_activity_ =
- AudioFrame::kVadPassive;
-
- EXPECT_CALL(output_receiver, NewMixedAudio(_, _, _, _))
- .Times(AtLeast(1));
-
- mixer->Process();
-
- for (int i = 0; i < kParticipants; ++i) {
- bool is_mixed = participants[i].IsMixed();
- if (i == kParticipants - 1 || i < kParticipants - 1 -
- AudioConferenceMixer::kMaximumAmountOfMixedParticipants) {
- EXPECT_FALSE(is_mixed) << "Mixing status of Participant #"
- << i << " wrong.";
- } else {
- EXPECT_TRUE(is_mixed) << "Mixing status of Participant #"
- << i << " wrong.";
- }
- }
-
- EXPECT_EQ(0, mixer->UnRegisterMixedStreamCallback());
-}
-
-} // namespace webrtc
diff --git a/modules/audio_mixer/DEPS b/modules/audio_mixer/DEPS
index a076950..51608ad 100644
--- a/modules/audio_mixer/DEPS
+++ b/modules/audio_mixer/DEPS
@@ -3,7 +3,6 @@
"+call",
"+common_audio",
"+modules/audio_coding",
- "+modules/audio_conference_mixer",
"+modules/audio_device",
"+modules/audio_processing",
"+modules/media_file",
diff --git a/test/mock_voe_channel_proxy.h b/test/mock_voe_channel_proxy.h
index 77fb171..0778bda 100644
--- a/test/mock_voe_channel_proxy.h
+++ b/test/mock_voe_channel_proxy.h
@@ -77,7 +77,7 @@
MOCK_METHOD2(GetAudioFrameWithInfo,
AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
AudioFrame* audio_frame));
- MOCK_CONST_METHOD0(NeededFrequency, int());
+ MOCK_CONST_METHOD0(PreferredSampleRate, int());
MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
MOCK_METHOD1(AssociateSendChannel,
void(const ChannelProxy& send_channel_proxy));
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index b9af474..444c5fd 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -19,8 +19,6 @@
"include/voe_base.h",
"include/voe_errors.h",
"monitor_module.h",
- "output_mixer.cc",
- "output_mixer.h",
"shared_data.cc",
"shared_data.h",
"statistics.cc",
@@ -74,7 +72,6 @@
"../modules:module_api",
"../modules/audio_coding:audio_format_conversion",
"../modules/audio_coding:rent_a_codec",
- "../modules/audio_conference_mixer",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/bitrate_controller",
@@ -109,7 +106,6 @@
"../common_audio",
"../modules:module_api",
"../modules/audio_coding",
- "../modules/audio_conference_mixer",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/media_file",
diff --git a/voice_engine/DEPS b/voice_engine/DEPS
index 496bf29..c8e9a1c 100644
--- a/voice_engine/DEPS
+++ b/voice_engine/DEPS
@@ -4,7 +4,6 @@
"+common_audio",
"+logging/rtc_event_log",
"+modules/audio_coding",
- "+modules/audio_conference_mixer",
"+modules/audio_device",
"+modules/audio_processing",
"+modules/media_file",
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 85b14fd..aaca65a 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -39,7 +39,6 @@
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/trace.h"
-#include "voice_engine/output_mixer.h"
#include "voice_engine/statistics.h"
#include "voice_engine/utility.h"
@@ -619,15 +618,17 @@
return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
}
-MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
- int32_t id,
- AudioFrame* audioFrame) {
+AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
+ int sample_rate_hz,
+ AudioFrame* audio_frame) {
+ audio_frame->sample_rate_hz_ = sample_rate_hz;
+
unsigned int ssrc;
RTC_CHECK_EQ(GetRemoteSSRC(ssrc), 0);
event_log_proxy_->LogAudioPlayout(ssrc);
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
bool muted;
- if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
+ if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame,
&muted) == -1) {
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::GetAudioFrame() PlayoutData10Ms() failed!");
@@ -635,20 +636,20 @@
// error so that the audio mixer module doesn't add it to the mix. As
// a result, it won't be played out and the actions skipped here are
// irrelevant.
- return MixerParticipant::AudioFrameInfo::kError;
+ return AudioMixer::Source::AudioFrameInfo::kError;
}
if (muted) {
// TODO(henrik.lundin): We should be able to do better than this. But we
// will have to go through all the cases below where the audio samples may
// be used, and handle the muted case in some way.
- AudioFrameOperations::Mute(audioFrame);
+ AudioFrameOperations::Mute(audio_frame);
}
// Convert module ID to internal VoE channel ID
- audioFrame->id_ = VoEChannelId(audioFrame->id_);
+ audio_frame->id_ = VoEChannelId(audio_frame->id_);
// Store speech type for dead-or-alive detection
- _outputSpeechType = audioFrame->speech_type_;
+ _outputSpeechType = audio_frame->speech_type_;
{
// Pass the audio buffers to an optional sink callback, before applying
@@ -658,9 +659,9 @@
rtc::CritScope cs(&_callbackCritSect);
if (audio_sink_) {
AudioSinkInterface::Data data(
- audioFrame->data(), audioFrame->samples_per_channel_,
- audioFrame->sample_rate_hz_, audioFrame->num_channels_,
- audioFrame->timestamp_);
+ audio_frame->data(), audio_frame->samples_per_channel_,
+ audio_frame->sample_rate_hz_, audio_frame->num_channels_,
+ audio_frame->timestamp_);
audio_sink_->OnData(data);
}
}
@@ -674,89 +675,53 @@
// Output volume scaling
if (output_gain < 0.99f || output_gain > 1.01f) {
// TODO(solenberg): Combine with mute state - this can cause clicks!
- AudioFrameOperations::ScaleWithSat(output_gain, audioFrame);
+ AudioFrameOperations::ScaleWithSat(output_gain, audio_frame);
}
// Measure audio level (0-9)
// TODO(henrik.lundin) Use the |muted| information here too.
// TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| (see
// https://crbug.com/webrtc/7517).
- _outputAudioLevel.ComputeLevel(*audioFrame, kAudioSampleDurationSeconds);
+ _outputAudioLevel.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds);
- if (capture_start_rtp_time_stamp_ < 0 && audioFrame->timestamp_ != 0) {
+ if (capture_start_rtp_time_stamp_ < 0 && audio_frame->timestamp_ != 0) {
// The first frame with a valid rtp timestamp.
- capture_start_rtp_time_stamp_ = audioFrame->timestamp_;
+ capture_start_rtp_time_stamp_ = audio_frame->timestamp_;
}
if (capture_start_rtp_time_stamp_ >= 0) {
- // audioFrame.timestamp_ should be valid from now on.
+ // audio_frame.timestamp_ should be valid from now on.
// Compute elapsed time.
int64_t unwrap_timestamp =
- rtp_ts_wraparound_handler_->Unwrap(audioFrame->timestamp_);
- audioFrame->elapsed_time_ms_ =
+ rtp_ts_wraparound_handler_->Unwrap(audio_frame->timestamp_);
+ audio_frame->elapsed_time_ms_ =
(unwrap_timestamp - capture_start_rtp_time_stamp_) /
(GetRtpTimestampRateHz() / 1000);
{
rtc::CritScope lock(&ts_stats_lock_);
// Compute ntp time.
- audioFrame->ntp_time_ms_ =
- ntp_estimator_.Estimate(audioFrame->timestamp_);
+ audio_frame->ntp_time_ms_ =
+ ntp_estimator_.Estimate(audio_frame->timestamp_);
// |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received.
- if (audioFrame->ntp_time_ms_ > 0) {
+ if (audio_frame->ntp_time_ms_ > 0) {
// Compute |capture_start_ntp_time_ms_| so that
// |capture_start_ntp_time_ms_| + |elapsed_time_ms_| == |ntp_time_ms_|
capture_start_ntp_time_ms_ =
- audioFrame->ntp_time_ms_ - audioFrame->elapsed_time_ms_;
+ audio_frame->ntp_time_ms_ - audio_frame->elapsed_time_ms_;
}
}
}
- return muted ? MixerParticipant::AudioFrameInfo::kMuted
- : MixerParticipant::AudioFrameInfo::kNormal;
+ return muted ? AudioMixer::Source::AudioFrameInfo::kMuted
+ : AudioMixer::Source::AudioFrameInfo::kNormal;
}
-AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
- int sample_rate_hz,
- AudioFrame* audio_frame) {
- audio_frame->sample_rate_hz_ = sample_rate_hz;
-
- const auto frame_info = GetAudioFrameWithMuted(-1, audio_frame);
-
- using FrameInfo = AudioMixer::Source::AudioFrameInfo;
- FrameInfo new_audio_frame_info = FrameInfo::kError;
- switch (frame_info) {
- case MixerParticipant::AudioFrameInfo::kNormal:
- new_audio_frame_info = FrameInfo::kNormal;
- break;
- case MixerParticipant::AudioFrameInfo::kMuted:
- new_audio_frame_info = FrameInfo::kMuted;
- break;
- case MixerParticipant::AudioFrameInfo::kError:
- new_audio_frame_info = FrameInfo::kError;
- break;
- }
- return new_audio_frame_info;
-}
-
-int32_t Channel::NeededFrequency(int32_t id) const {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::NeededFrequency(id=%d)", id);
-
- int highestNeeded = 0;
-
- // Determine highest needed receive frequency
- int32_t receiveFrequency = audio_coding_->ReceiveFrequency();
-
+int Channel::PreferredSampleRate() const {
// Return the bigger of playout and receive frequency in the ACM.
- if (audio_coding_->PlayoutFrequency() > receiveFrequency) {
- highestNeeded = audio_coding_->PlayoutFrequency();
- } else {
- highestNeeded = receiveFrequency;
- }
-
- return highestNeeded;
+ return std::max(audio_coding_->ReceiveFrequency(),
+ audio_coding_->PlayoutFrequency());
}
int32_t Channel::CreateChannel(Channel*& channel,
@@ -806,7 +771,6 @@
capture_start_rtp_time_stamp_(-1),
capture_start_ntp_time_ms_(-1),
_engineStatisticsPtr(NULL),
- _outputMixerPtr(NULL),
_moduleProcessThreadPtr(NULL),
_audioDeviceModulePtr(NULL),
_voiceEngineObserverPtr(NULL),
@@ -983,7 +947,6 @@
}
int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
- OutputMixer& outputMixer,
ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
VoiceEngineObserver* voiceEngineObserver,
@@ -994,7 +957,6 @@
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetEngineInformation()");
_engineStatisticsPtr = &engineStatistics;
- _outputMixerPtr = &outputMixer;
_moduleProcessThreadPtr = &moduleProcessThread;
_audioDeviceModulePtr = &audioDeviceModule;
_voiceEngineObserverPtr = voiceEngineObserver;
@@ -1020,14 +982,6 @@
return 0;
}
- // Add participant as candidates for mixing.
- if (_outputMixerPtr->SetMixabilityStatus(*this, true) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StartPlayout() failed to add participant to mixer");
- return -1;
- }
-
channel_state_.SetPlaying(true);
return 0;
@@ -1040,14 +994,6 @@
return 0;
}
- // Remove participant as candidates for mixing
- if (_outputMixerPtr->SetMixabilityStatus(*this, false) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StopPlayout() failed to remove participant from mixer");
- return -1;
- }
-
channel_state_.SetPlaying(false);
_outputAudioLevel.Clear();
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index f4e47ee..53b8cd0 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -23,7 +23,6 @@
#include "modules/audio_coding/acm2/codec_manager.h"
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/audio_coding/include/audio_coding_module.h"
-#include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "modules/audio_processing/rms_level.h"
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
@@ -89,7 +88,6 @@
namespace voe {
-class OutputMixer;
class RtcEventLogProxy;
class RtcpRttStatsProxy;
class RtpPacketSenderProxy;
@@ -144,7 +142,6 @@
public Transport,
public AudioPacketizationCallback, // receive encoded packets from the
// ACM
- public MixerParticipant, // supplies output mixer with audio frames
public OverheadObserver {
public:
friend class VoERtcpObserver;
@@ -162,7 +159,6 @@
int32_t Init();
void Terminate();
int32_t SetEngineInformation(Statistics& engineStatistics,
- OutputMixer& outputMixer,
ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
VoiceEngineObserver* voiceEngineObserver,
@@ -283,17 +279,13 @@
const PacketOptions& packet_options) override;
bool SendRtcp(const uint8_t* data, size_t len) override;
- // From MixerParticipant
- MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted(
- int32_t id,
- AudioFrame* audioFrame) override;
- int32_t NeededFrequency(int32_t id) const override;
-
// From AudioMixer::Source.
AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame);
+ int PreferredSampleRate() const;
+
uint32_t InstanceId() const { return _instanceId; }
int32_t ChannelId() const { return _channelId; }
bool Playing() const { return channel_state_.Get().playing; }
@@ -433,7 +425,6 @@
// uses
Statistics* _engineStatisticsPtr;
- OutputMixer* _outputMixerPtr;
ProcessThread* _moduleProcessThreadPtr;
AudioDeviceModule* _audioDeviceModulePtr;
VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base
diff --git a/voice_engine/channel_proxy.cc b/voice_engine/channel_proxy.cc
index f6013c2..15db60d 100644
--- a/voice_engine/channel_proxy.cc
+++ b/voice_engine/channel_proxy.cc
@@ -257,9 +257,9 @@
return channel()->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
}
-int ChannelProxy::NeededFrequency() const {
+int ChannelProxy::PreferredSampleRate() const {
RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
- return static_cast<int>(channel()->NeededFrequency(-1));
+ return channel()->PreferredSampleRate();
}
void ChannelProxy::SetTransportOverhead(int transport_overhead_per_packet) {
diff --git a/voice_engine/channel_proxy.h b/voice_engine/channel_proxy.h
index 40246aaa..cb292ec 100644
--- a/voice_engine/channel_proxy.h
+++ b/voice_engine/channel_proxy.h
@@ -107,7 +107,7 @@
virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame);
- virtual int NeededFrequency() const;
+ virtual int PreferredSampleRate() const;
virtual void SetTransportOverhead(int transport_overhead_per_packet);
virtual void AssociateSendChannel(const ChannelProxy& send_channel_proxy);
virtual void DisassociateSendChannel();
diff --git a/voice_engine/output_mixer.cc b/voice_engine/output_mixer.cc
deleted file mode 100644
index 8d1e31d..0000000
--- a/voice_engine/output_mixer.cc
+++ /dev/null
@@ -1,157 +0,0 @@
-/*
- * 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 "voice_engine/output_mixer.h"
-
-#include "modules/audio_processing/include/audio_processing.h"
-#include "rtc_base/format_macros.h"
-#include "system_wrappers/include/trace.h"
-#include "voice_engine/statistics.h"
-#include "voice_engine/utility.h"
-
-namespace webrtc {
-namespace voe {
-
-void
-OutputMixer::NewMixedAudio(int32_t id,
- const AudioFrame& generalAudioFrame,
- const AudioFrame** uniqueAudioFrames,
- uint32_t size)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::NewMixedAudio(id=%d, size=%u)", id, size);
-
- _audioFrame.CopyFrom(generalAudioFrame);
- _audioFrame.id_ = id;
-}
-
-int32_t
-OutputMixer::Create(OutputMixer*& mixer, uint32_t instanceId)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
- "OutputMixer::Create(instanceId=%d)", instanceId);
- mixer = new OutputMixer(instanceId);
- if (mixer == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
- "OutputMixer::Create() unable to allocate memory for"
- "mixer");
- return -1;
- }
- return 0;
-}
-
-OutputMixer::OutputMixer(uint32_t instanceId) :
- _mixerModule(*AudioConferenceMixer::Create(instanceId)),
- _instanceId(instanceId),
- _mixingFrequencyHz(8000)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::OutputMixer() - ctor");
-
- if (_mixerModule.RegisterMixedStreamCallback(this) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::OutputMixer() failed to register mixer"
- "callbacks");
- }
-}
-
-void
-OutputMixer::Destroy(OutputMixer*& mixer)
-{
- if (mixer)
- {
- delete mixer;
- mixer = NULL;
- }
-}
-
-OutputMixer::~OutputMixer()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::~OutputMixer() - dtor");
- _mixerModule.UnRegisterMixedStreamCallback();
- delete &_mixerModule;
-}
-
-int32_t
-OutputMixer::SetEngineInformation(voe::Statistics& engineStatistics)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::SetEngineInformation()");
- _engineStatisticsPtr = &engineStatistics;
- return 0;
-}
-
-int32_t
-OutputMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::SetAudioProcessingModule("
- "audioProcessingModule=0x%x)", audioProcessingModule);
- _audioProcessingModulePtr = audioProcessingModule;
- return 0;
-}
-
-int32_t
-OutputMixer::SetMixabilityStatus(MixerParticipant& participant,
- bool mixable)
-{
- return _mixerModule.SetMixabilityStatus(&participant, mixable);
-}
-
-int32_t
-OutputMixer::MixActiveChannels()
-{
- _mixerModule.Process();
- return 0;
-}
-
-int OutputMixer::GetMixedAudio(int sample_rate_hz,
- size_t num_channels,
- AudioFrame* frame) {
- WEBRTC_TRACE(
- kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%" PRIuS ")",
- sample_rate_hz, num_channels);
-
- 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
-OutputMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm)
-{
- if (_audioFrame.sample_rate_hz_ != _mixingFrequencyHz)
- {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::DoOperationsOnCombinedSignal() => "
- "mixing frequency = %d", _audioFrame.sample_rate_hz_);
- _mixingFrequencyHz = _audioFrame.sample_rate_hz_;
- }
-
- // --- 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_NOTREACHED();
- }
- }
-
- return 0;
-}
-} // namespace voe
-} // namespace webrtc
diff --git a/voice_engine/output_mixer.h b/voice_engine/output_mixer.h
deleted file mode 100644
index 284f92d..0000000
--- a/voice_engine/output_mixer.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * 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 VOICE_ENGINE_OUTPUT_MIXER_H_
-#define VOICE_ENGINE_OUTPUT_MIXER_H_
-
-#include <memory>
-
-#include "common_audio/resampler/include/push_resampler.h"
-#include "common_types.h" // NOLINT(build/include)
-#include "modules/audio_conference_mixer/include/audio_conference_mixer.h"
-#include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
-#include "rtc_base/criticalsection.h"
-
-namespace webrtc {
-
-class AudioProcessing;
-class FileWrapper;
-
-namespace voe {
-
-class Statistics;
-
-class OutputMixer : public AudioMixerOutputReceiver
-{
-public:
- static int32_t Create(OutputMixer*& mixer, uint32_t instanceId);
-
- static void Destroy(OutputMixer*& mixer);
-
- int32_t SetEngineInformation(Statistics& engineStatistics);
-
- int32_t SetAudioProcessingModule(
- AudioProcessing* audioProcessingModule);
-
- int32_t MixActiveChannels();
-
- int32_t DoOperationsOnCombinedSignal(bool feed_data_to_apm);
-
- int32_t SetMixabilityStatus(MixerParticipant& participant,
- bool mixable);
-
- int GetMixedAudio(int sample_rate_hz, size_t num_channels,
- AudioFrame* audioFrame);
-
- virtual ~OutputMixer();
-
- // from AudioMixerOutputReceiver
- virtual void NewMixedAudio(
- int32_t id,
- const AudioFrame& generalAudioFrame,
- const AudioFrame** uniqueAudioFrames,
- uint32_t size);
-
-private:
- OutputMixer(uint32_t instanceId);
-
- // uses
- Statistics* _engineStatisticsPtr;
- AudioProcessing* _audioProcessingModulePtr;
-
- AudioConferenceMixer& _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_;
- int _instanceId;
- int _mixingFrequencyHz;
-};
-
-} // namespace voe
-
-} // namespace werbtc
-
-#endif // VOICE_ENGINE_OUTPUT_MIXER_H_
diff --git a/voice_engine/shared_data.cc b/voice_engine/shared_data.cc
index dc3d8c0..3b3f17f 100644
--- a/voice_engine/shared_data.cc
+++ b/voice_engine/shared_data.cc
@@ -13,7 +13,6 @@
#include "modules/audio_processing/include/audio_processing.h"
#include "system_wrappers/include/trace.h"
#include "voice_engine/channel.h"
-#include "voice_engine/output_mixer.h"
#include "voice_engine/transmit_mixer.h"
namespace webrtc {
@@ -30,9 +29,6 @@
_moduleProcessThreadPtr(ProcessThread::Create("VoiceProcessThread")),
encoder_queue_("AudioEncoderQueue") {
Trace::CreateTrace();
- if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0) {
- _outputMixerPtr->SetEngineInformation(_engineStatistics);
- }
if (TransmitMixer::Create(_transmitMixerPtr, _gInstanceCounter) == 0) {
_transmitMixerPtr->SetEngineInformation(*_moduleProcessThreadPtr,
_engineStatistics, _channelManager);
@@ -41,7 +37,6 @@
SharedData::~SharedData()
{
- OutputMixer::Destroy(_outputMixerPtr);
TransmitMixer::Destroy(_transmitMixerPtr);
if (_audioDevicePtr) {
_audioDevicePtr->Release();
@@ -62,7 +57,6 @@
void SharedData::set_audio_processing(AudioProcessing* audioproc) {
_transmitMixerPtr->SetAudioProcessingModule(audioproc);
- _outputMixerPtr->SetAudioProcessingModule(audioproc);
}
int SharedData::NumOfSendingChannels() {
diff --git a/voice_engine/shared_data.h b/voice_engine/shared_data.h
index 58ac827..336aab2 100644
--- a/voice_engine/shared_data.h
+++ b/voice_engine/shared_data.h
@@ -31,7 +31,6 @@
namespace voe {
class TransmitMixer;
-class OutputMixer;
class SharedData
{
@@ -45,7 +44,6 @@
const rtc::scoped_refptr<AudioDeviceModule>& audio_device);
void set_audio_processing(AudioProcessing* audio_processing);
TransmitMixer* transmit_mixer() { return _transmitMixerPtr; }
- OutputMixer* output_mixer() { return _outputMixerPtr; }
rtc::CriticalSection* crit_sec() { return &_apiCritPtr; }
ProcessThread* process_thread() { return _moduleProcessThreadPtr.get(); }
rtc::TaskQueue* encoder_queue();
@@ -66,7 +64,6 @@
ChannelManager _channelManager;
Statistics _engineStatistics;
rtc::scoped_refptr<AudioDeviceModule> _audioDevicePtr;
- OutputMixer* _outputMixerPtr;
TransmitMixer* _transmitMixerPtr;
std::unique_ptr<ProcessThread> _moduleProcessThreadPtr;
// |encoder_queue| is defined last to ensure all pending tasks are cancelled
diff --git a/voice_engine/voe_base_impl.cc b/voice_engine/voe_base_impl.cc
index ec28e06..76dd55a 100644
--- a/voice_engine/voe_base_impl.cc
+++ b/voice_engine/voe_base_impl.cc
@@ -18,12 +18,9 @@
#include "rtc_base/format_macros.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
-#include "system_wrappers/include/file_wrapper.h"
#include "voice_engine/channel.h"
#include "voice_engine/include/voe_errors.h"
-#include "voice_engine/output_mixer.h"
#include "voice_engine/transmit_mixer.h"
-#include "voice_engine/utility.h"
#include "voice_engine/voice_engine_impl.h"
namespace webrtc {
@@ -148,9 +145,7 @@
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
- GetPlayoutData(static_cast<int>(samplesPerSec), nChannels, nSamples, true,
- audioSamples, elapsed_time_ms, ntp_time_ms);
- nSamplesOut = audioFrame_.samples_per_channel_;
+ RTC_NOTREACHED();
return 0;
}
@@ -177,11 +172,7 @@
size_t number_of_frames,
void* audio_data, int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
- assert(bits_per_sample == 16);
- assert(number_of_frames == static_cast<size_t>(sample_rate / 100));
-
- GetPlayoutData(sample_rate, number_of_channels, number_of_frames, false,
- audio_data, elapsed_time_ms, ntp_time_ms);
+ RTC_NOTREACHED();
}
int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
@@ -418,7 +409,7 @@
int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
if (channel_owner->channel()->SetEngineInformation(
- shared_->statistics(), *shared_->output_mixer(),
+ shared_->statistics(),
*shared_->process_thread(), *shared_->audio_device(),
voiceEngineObserverPtr_, &callbackCritSect_,
shared_->encoder_queue()) != 0) {
@@ -653,34 +644,4 @@
return shared_->statistics().SetUnInitialized();
}
-
-void VoEBaseImpl::GetPlayoutData(int sample_rate, size_t number_of_channels,
- size_t number_of_frames, bool feed_data_to_apm,
- void* audio_data, int64_t* elapsed_time_ms,
- int64_t* ntp_time_ms) {
- assert(shared_->output_mixer() != nullptr);
-
- // TODO(andrew): if the device is running in mono, we should tell the mixer
- // here so that it will only request mono from AudioCodingModule.
- // Perform mixing of all active participants (channel-based mixing)
- shared_->output_mixer()->MixActiveChannels();
-
- // Additional operations on the combined signal
- shared_->output_mixer()->DoOperationsOnCombinedSignal(feed_data_to_apm);
-
- // Retrieve the final output mix (resampled to match the ADM)
- shared_->output_mixer()->GetMixedAudio(sample_rate, number_of_channels,
- &audioFrame_);
-
- assert(number_of_frames == audioFrame_.samples_per_channel_);
- assert(sample_rate == audioFrame_.sample_rate_hz_);
-
- // Deliver audio (PCM) samples to the ADM
- memcpy(audio_data, audioFrame_.data(),
- sizeof(int16_t) * number_of_frames * number_of_channels);
-
- *elapsed_time_ms = audioFrame_.elapsed_time_ms_;
- *ntp_time_ms = audioFrame_.ntp_time_ms_;
-}
-
} // namespace webrtc
diff --git a/voice_engine/voe_base_impl.h b/voice_engine/voe_base_impl.h
index 608292c..e15d1f1 100644
--- a/voice_engine/voe_base_impl.h
+++ b/voice_engine/voe_base_impl.h
@@ -62,27 +62,27 @@
const uint32_t volume,
const bool key_pressed,
uint32_t& new_mic_volume) override;
- int32_t NeedMorePlayData(const size_t nSamples,
- const size_t nBytesPerSample,
- const size_t nChannels,
- const uint32_t samplesPerSec,
- void* audioSamples,
- size_t& nSamplesOut,
- int64_t* elapsed_time_ms,
- int64_t* ntp_time_ms) override;
+ RTC_DEPRECATED int32_t NeedMorePlayData(const size_t nSamples,
+ const size_t nBytesPerSample,
+ const size_t nChannels,
+ const uint32_t samplesPerSec,
+ void* audioSamples,
+ size_t& nSamplesOut,
+ int64_t* elapsed_time_ms,
+ int64_t* ntp_time_ms) override;
void PushCaptureData(int voe_channel,
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override;
- void PullRenderData(int bits_per_sample,
- int sample_rate,
- size_t number_of_channels,
- size_t number_of_frames,
- void* audio_data,
- int64_t* elapsed_time_ms,
- int64_t* ntp_time_ms) override;
+ RTC_DEPRECATED void PullRenderData(int bits_per_sample,
+ int sample_rate,
+ size_t number_of_channels,
+ size_t number_of_frames,
+ void* audio_data,
+ int64_t* elapsed_time_ms,
+ int64_t* ntp_time_ms) override;
// AudioDeviceObserver
void OnErrorIsReported(const ErrorCode error) override;
