Nack for audio.
R=stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1507004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4188 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -8,14 +8,13 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/main/test/Channel.h"
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#include <assert.h>
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#include <iostream>
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#include "audio_coding_module.h"
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#include "Channel.h"
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#include "tick_util.h"
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#include "typedefs.h"
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#include "common_types.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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namespace webrtc {
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@ -29,9 +28,12 @@ int32_t Channel::SendData(const FrameType frameType, const uint8_t payloadType,
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rtpInfo.header.markerBit = false;
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rtpInfo.header.ssrc = 0;
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rtpInfo.header.sequenceNumber = _seqNo++;
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rtpInfo.header.sequenceNumber = (external_sequence_number_ < 0) ?
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_seqNo++ : static_cast<uint16_t>(external_sequence_number_);
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rtpInfo.header.payloadType = payloadType;
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rtpInfo.header.timestamp = timeStamp;
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rtpInfo.header.timestamp = (external_send_timestamp_ < 0) ? timeStamp :
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static_cast<uint32_t>(external_send_timestamp_);
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if (frameType == kAudioFrameCN) {
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rtpInfo.type.Audio.isCNG = true;
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} else {
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@ -108,6 +110,11 @@ int32_t Channel::SendData(const FrameType frameType, const uint8_t payloadType,
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}
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}
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if (num_packets_to_drop_ > 0) {
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num_packets_to_drop_--;
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return 0;
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}
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status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtpInfo);
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return status;
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@ -208,7 +215,10 @@ Channel::Channel(int16_t chID)
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_packetLoss(0),
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_useFECTestWithPacketLoss(false),
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_beginTime(TickTime::MillisecondTimestamp()),
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_totalBytes(0) {
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_totalBytes(0),
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external_send_timestamp_(-1),
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external_sequence_number_(-1),
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num_packets_to_drop_(0) {
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int n;
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int k;
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for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
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