Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 77 insertions, 0 deletions

diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
new file mode 100644
index 0000000000..7ff39579ee
--- /dev/null
+++ b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+
+#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/interface/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;
+    int32_t 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 // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_