author Randell Jesup <>
Wed, 18 Nov 2015 15:03:25 -0500
changeset 307280 ae32ad44ce1c048258a881dbaa7727def08c53a4
parent 307279 f62c9e49a44fb473b5c8701776d8f5ae3da22cbc
child 372405 e10e9f0e5ca20b500efb59dc5e4c25f248692b96
permissions -rw-r--r--
Bug 1198458: Rollup of changes previously applied to media/webrtc/trunk/webrtc and fixes to those rs=jesup r=froyd,jib,bwc,jesup,gcp,sotaro,pkerr,pehrsons Landing as one rolled-up patch to avoid breaking regression tests, and in keeping with previous WebRTC imports. Broken out parts that needed review are on the bug.

 *  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.

// This sub-API supports the following functionalities:
//  - RTP header modification (time stamp and sequence number fields).
//  - Playout delay tuning to synchronize the voice with video.
//  - Playout delay monitoring.
// Usage example, omitting error checking:
//  using namespace webrtc;
//  VoiceEngine* voe = VoiceEngine::Create();
//  VoEBase* base = VoEBase::GetInterface(voe);
//  VoEVideoSync* vsync  = VoEVideoSync::GetInterface(voe);
//  base->Init();
//  ...
//  int buffer_ms(0);
//  vsync->GetPlayoutBufferSize(buffer_ms);
//  ...
//  base->Terminate();
//  base->Release();
//  vsync->Release();
//  VoiceEngine::Delete(voe);

#include "webrtc/common_types.h"

namespace webrtc {

class RtpReceiver;
class RtpRtcp;
class VoiceEngine;

    // Factory for the VoEVideoSync sub-API. Increases an internal
    // reference counter if successful. Returns NULL if the API is not
    // supported or if construction fails.
    static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine);

    // Releases the VoEVideoSync 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;

    // Gets the current sound card buffer size (playout delay).
    virtual int GetPlayoutBufferSize(int& buffer_ms) = 0;

    // Sets a minimum target delay for the jitter buffer. This delay is
    // maintained by the jitter buffer, unless channel condition (jitter in
    // inter-arrival times) dictates a higher required delay. The overall
    // jitter buffer delay is max of |delay_ms| and the latency that NetEq
    // computes based on inter-arrival times and its playout mode.
    virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0;

    // Sets the current a/v delay in ms (negative is video leading) if known,
    // otherwise 0.
    virtual int SetCurrentSyncOffset(int channel, int offset_ms) = 0;

    // Sets an initial delay for the playout jitter buffer. The playout of the
    // audio is delayed by |delay_ms| in milliseconds. Thereafter, the delay is
    // maintained, unless NetEq's internal mechanism requires a higher latency.
    // Such a latency is computed based on inter-arrival times and NetEq's
    // playout mode.
    virtual int SetInitialPlayoutDelay(int channel, int delay_ms) = 0;

    // Gets the |jitter_buffer_delay_ms| (including the algorithmic delay),
    // the |playout_buffer_delay_ms| and |avsync_offset_ms| for a specified
    // |channel|.
    virtual int GetDelayEstimate(int channel,
                                 int* jitter_buffer_delay_ms,
                                 int* playout_buffer_delay_ms,
                                 int* avsync_offset_ms) = 0;

    // Returns the least required jitter buffer delay. This is computed by the
    // the jitter buffer based on the inter-arrival time of RTP packets and
    // playout mode. NetEq maintains this latency unless a higher value is
    // requested by calling SetMinimumPlayoutDelay().
    virtual int GetLeastRequiredDelayMs(int channel) const = 0;

    // Manual initialization of the RTP timestamp.
    virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0;

    // Manual initialization of the RTP sequence number.
    virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0;

    // Get the received RTP timestamp
    virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0;

    virtual int GetRtpRtcp (int channel, RtpRtcp** rtpRtcpModule,
                            RtpReceiver** rtp_receiver) = 0;

    VoEVideoSync() { }
    virtual ~VoEVideoSync() { }

}  // namespace webrtc