Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

110 lines
3.5 KiB
C++

/*
* Copyright (c) 2012 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.
*/
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/defines.h"
#include "webrtc/system_wrappers/interface/logging.h"
namespace webrtc {
void TimestampScaler::ToInternal(Packet* packet) {
if (!packet) {
return;
}
packet->header.timestamp = ToInternal(packet->header.timestamp,
packet->header.payloadType);
}
void TimestampScaler::ToInternal(PacketList* packet_list) {
PacketList::iterator it;
for (it = packet_list->begin(); it != packet_list->end(); ++it) {
ToInternal(*it);
}
}
uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
uint8_t rtp_payload_type) {
const DecoderDatabase::DecoderInfo* info =
decoder_database_.GetDecoderInfo(rtp_payload_type);
if (!info) {
// Payload type is unknown. Do not scale.
return external_timestamp;
}
switch (info->codec_type) {
case kDecoderG722:
case kDecoderG722_2ch: {
// Use timestamp scaling with factor 2 (two output samples per RTP
// timestamp).
numerator_ = 2;
denominator_ = 1;
break;
}
case kDecoderISACfb:
case kDecoderCNGswb48kHz: {
// Use timestamp scaling with factor 2/3 (32 kHz sample rate, but RTP
// timestamps run on 48 kHz).
// TODO(tlegrand): Remove scaling for kDecoderCNGswb48kHz once ACM has
// full 48 kHz support.
numerator_ = 2;
denominator_ = 3;
}
case kDecoderAVT:
case kDecoderCNGnb:
case kDecoderCNGwb:
case kDecoderCNGswb32kHz: {
// Do not change the timestamp scaling settings for DTMF or CNG.
break;
}
default: {
// Do not use timestamp scaling for any other codec.
numerator_ = 1;
denominator_ = 1;
break;
}
}
if (!(numerator_ == 1 && denominator_ == 1)) {
// We have a scale factor != 1.
if (!first_packet_received_) {
external_ref_ = external_timestamp;
internal_ref_ = external_timestamp;
first_packet_received_ = true;
}
int32_t external_diff = external_timestamp - external_ref_;
assert(denominator_ > 0); // Should not be possible.
external_ref_ = external_timestamp;
internal_ref_ += (external_diff * numerator_) / denominator_;
LOG(LS_VERBOSE) << "Converting timestamp: " << external_timestamp <<
" -> " << internal_ref_;
return internal_ref_;
} else {
// No scaling.
return external_timestamp;
}
}
uint32_t TimestampScaler::ToExternal(uint32_t internal_timestamp) const {
if (!first_packet_received_ || (numerator_ == 1 && denominator_ == 1)) {
// Not initialized, or scale factor is 1.
return internal_timestamp;
} else {
int32_t internal_diff = internal_timestamp - internal_ref_;
assert(numerator_ > 0); // Should not be possible.
// Do not update references in this method.
// Switch |denominator_| and |numerator_| to convert the other way.
return external_ref_ + (internal_diff * denominator_) / numerator_;
}
}
} // namespace webrtc