blob: cf16d3b3cf8e6772bd5d740a67955d9d48579109 [file] [log] [blame]
* 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 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), and
// the |playout_buffer_delay_ms| for a specified |channel|.
virtual int GetDelayEstimate(int channel,
int* jitter_buffer_delay_ms,
int* playout_buffer_delay_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