a9cc40b7d2
[This CL is a rebase of an original CL by solenberg@: https://codereview.webrtc.org/2948763002/ which in turn was a rebase of an original CL by peah@: https://chromium-review.googlesource.com/c/527032/] Allow an external audio processing module to be used in WebRTC This CL adds support for optionally using an externally created audio processing module in a peerconnection. The ownership is shared between the peerconnection and the external creator of the module. As part of this the internal ownership of the audio processing module is moved from VoiceEngine to WebRtcVoiceEngine. BUG=webrtc:7775 Review-Url: https://codereview.webrtc.org/2961723004 Cr-Commit-Position: refs/heads/master@{#18837}
105 lines
3.9 KiB
C++
105 lines
3.9 KiB
C++
/*
|
|
* 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_MEDIA_ENGINE_WEBRTCMEDIAENGINE_H_
|
|
#define WEBRTC_MEDIA_ENGINE_WEBRTCMEDIAENGINE_H_
|
|
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "webrtc/call/call.h"
|
|
#include "webrtc/config.h"
|
|
#include "webrtc/media/base/mediaengine.h"
|
|
|
|
namespace webrtc {
|
|
class AudioDecoderFactory;
|
|
class AudioDeviceModule;
|
|
class AudioMixer;
|
|
class AudioProcessing;
|
|
}
|
|
namespace cricket {
|
|
class WebRtcVideoDecoderFactory;
|
|
class WebRtcVideoEncoderFactory;
|
|
}
|
|
|
|
namespace cricket {
|
|
|
|
class WebRtcMediaEngineFactory {
|
|
public:
|
|
// These Create methods may be called on any thread, though the engine is
|
|
// only expected to be used on one thread, internally called the "worker
|
|
// thread". This is the thread Init must be called on.
|
|
|
|
// TODO(ossu): Backwards-compatible interface. Will be deprecated once the
|
|
// audio decoder factory is fully plumbed and used throughout WebRTC.
|
|
// See: crbug.com/webrtc/6000
|
|
static MediaEngineInterface* Create(
|
|
webrtc::AudioDeviceModule* adm,
|
|
WebRtcVideoEncoderFactory* video_encoder_factory,
|
|
WebRtcVideoDecoderFactory* video_decoder_factory);
|
|
|
|
// TODO(deadbeef): Change these to return an std::unique_ptr<>, to indicate
|
|
// that the caller owns the returned object.
|
|
static MediaEngineInterface* Create(
|
|
webrtc::AudioDeviceModule* adm,
|
|
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
|
|
audio_decoder_factory,
|
|
WebRtcVideoEncoderFactory* video_encoder_factory,
|
|
WebRtcVideoDecoderFactory* video_decoder_factory);
|
|
|
|
static MediaEngineInterface* Create(
|
|
webrtc::AudioDeviceModule* adm,
|
|
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
|
|
audio_decoder_factory,
|
|
WebRtcVideoEncoderFactory* video_encoder_factory,
|
|
WebRtcVideoDecoderFactory* video_decoder_factory,
|
|
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
|
|
rtc::scoped_refptr<webrtc::AudioProcessing> apm);
|
|
|
|
static MediaEngineInterface* Create(
|
|
webrtc::AudioDeviceModule* adm,
|
|
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
|
|
audio_encoder_factory,
|
|
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
|
|
audio_decoder_factory,
|
|
WebRtcVideoEncoderFactory* video_encoder_factory,
|
|
WebRtcVideoDecoderFactory* video_decoder_factory);
|
|
|
|
static MediaEngineInterface* Create(
|
|
webrtc::AudioDeviceModule* adm,
|
|
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
|
|
audio_encoder_factory,
|
|
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
|
|
audio_decoder_factory,
|
|
WebRtcVideoEncoderFactory* video_encoder_factory,
|
|
WebRtcVideoDecoderFactory* video_decoder_factory,
|
|
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
|
|
rtc::scoped_refptr<webrtc::AudioProcessing> apm);
|
|
};
|
|
|
|
// Verify that extension IDs are within 1-byte extension range and are not
|
|
// overlapping.
|
|
bool ValidateRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions);
|
|
|
|
// Discard any extensions not validated by the 'supported' predicate. Duplicate
|
|
// extensions are removed if 'filter_redundant_extensions' is set, and also any
|
|
// mutually exclusive extensions (see implementation for details) are removed.
|
|
std::vector<webrtc::RtpExtension> FilterRtpExtensions(
|
|
const std::vector<webrtc::RtpExtension>& extensions,
|
|
bool (*supported)(const std::string&),
|
|
bool filter_redundant_extensions);
|
|
|
|
webrtc::Call::Config::BitrateConfig GetBitrateConfigForCodec(
|
|
const Codec& codec);
|
|
|
|
} // namespace cricket
|
|
|
|
#endif // WEBRTC_MEDIA_ENGINE_WEBRTCMEDIAENGINE_H_
|