webrtc_m130/webrtc/voice_engine/include/voe_external_media.h
Peter Kasting dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00

86 lines
3.3 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_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
#define WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
#include "webrtc/common_types.h"
namespace webrtc {
class VoiceEngine;
class AudioFrame;
class WEBRTC_DLLEXPORT VoEMediaProcess {
public:
// The VoiceEngine user should override the Process() method in a
// derived class. Process() will be called when audio is ready to
// be processed. The audio can be accessed in several different modes
// given by the |type| parameter. The function should modify the
// original data and ensure that it is copied back to the |audio10ms|
// array. The number of samples in the frame cannot be changed.
// The sampling frequency will depend upon the codec used.
// If |isStereo| is true, audio10ms will contain 16-bit PCM data
// samples in interleaved stereo format (L0,R0,L1,R1,...).
virtual void Process(int channel,
ProcessingTypes type,
int16_t audio10ms[],
size_t length,
int samplingFreq,
bool isStereo) = 0;
protected:
virtual ~VoEMediaProcess() {}
};
class WEBRTC_DLLEXPORT VoEExternalMedia {
public:
// Factory for the VoEExternalMedia sub-API. Increases an internal
// reference counter if successful. Returns NULL if the API is not
// supported or if construction fails.
static VoEExternalMedia* GetInterface(VoiceEngine* voiceEngine);
// Releases the VoEExternalMedia sub-API and decreases an internal
// reference counter. Returns the new reference count. This value should
// be zero for all sub-API:s before the VoiceEngine object can be safely
// deleted.
virtual int Release() = 0;
// Installs a VoEMediaProcess derived instance and activates external
// media for the specified |channel| and |type|.
virtual int RegisterExternalMediaProcessing(
int channel,
ProcessingTypes type,
VoEMediaProcess& processObject) = 0;
// Removes the VoEMediaProcess derived instance and deactivates external
// media for the specified |channel| and |type|.
virtual int DeRegisterExternalMediaProcessing(int channel,
ProcessingTypes type) = 0;
// Pulls an audio frame from the specified |channel| for external mixing.
// If the |desired_sample_rate_hz| is 0, the signal will be returned with
// its native frequency, otherwise it will be resampled. Valid frequencies
// are 16, 22, 32, 44 or 48 kHz.
virtual int GetAudioFrame(int channel,
int desired_sample_rate_hz,
AudioFrame* frame) = 0;
// Sets the state of external mixing. Cannot be changed during playback.
virtual int SetExternalMixing(int channel, bool enable) = 0;
protected:
VoEExternalMedia() {}
virtual ~VoEExternalMedia() {}
};
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H