mirror of
https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
110 lines
3.5 KiB
C++
110 lines
3.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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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/neteq/timestamp_scaler.h"
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#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
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#include "webrtc/modules/audio_coding/neteq/defines.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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namespace webrtc {
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void TimestampScaler::ToInternal(Packet* packet) {
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if (!packet) {
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return;
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}
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packet->header.timestamp = ToInternal(packet->header.timestamp,
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packet->header.payloadType);
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}
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void TimestampScaler::ToInternal(PacketList* packet_list) {
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PacketList::iterator it;
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for (it = packet_list->begin(); it != packet_list->end(); ++it) {
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ToInternal(*it);
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}
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}
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uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
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uint8_t rtp_payload_type) {
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const DecoderDatabase::DecoderInfo* info =
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decoder_database_.GetDecoderInfo(rtp_payload_type);
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if (!info) {
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// Payload type is unknown. Do not scale.
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return external_timestamp;
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}
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switch (info->codec_type) {
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case kDecoderG722:
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case kDecoderG722_2ch: {
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// Use timestamp scaling with factor 2 (two output samples per RTP
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// timestamp).
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numerator_ = 2;
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denominator_ = 1;
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break;
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}
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case kDecoderISACfb:
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case kDecoderCNGswb48kHz: {
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// Use timestamp scaling with factor 2/3 (32 kHz sample rate, but RTP
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// timestamps run on 48 kHz).
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// TODO(tlegrand): Remove scaling for kDecoderCNGswb48kHz once ACM has
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// full 48 kHz support.
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numerator_ = 2;
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denominator_ = 3;
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}
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case kDecoderAVT:
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case kDecoderCNGnb:
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case kDecoderCNGwb:
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case kDecoderCNGswb32kHz: {
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// Do not change the timestamp scaling settings for DTMF or CNG.
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break;
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}
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default: {
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// Do not use timestamp scaling for any other codec.
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numerator_ = 1;
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denominator_ = 1;
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break;
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}
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}
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if (!(numerator_ == 1 && denominator_ == 1)) {
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// We have a scale factor != 1.
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if (!first_packet_received_) {
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external_ref_ = external_timestamp;
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internal_ref_ = external_timestamp;
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first_packet_received_ = true;
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}
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int32_t external_diff = external_timestamp - external_ref_;
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assert(denominator_ > 0); // Should not be possible.
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external_ref_ = external_timestamp;
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internal_ref_ += (external_diff * numerator_) / denominator_;
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LOG(LS_VERBOSE) << "Converting timestamp: " << external_timestamp <<
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" -> " << internal_ref_;
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return internal_ref_;
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} else {
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// No scaling.
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return external_timestamp;
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}
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}
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uint32_t TimestampScaler::ToExternal(uint32_t internal_timestamp) const {
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if (!first_packet_received_ || (numerator_ == 1 && denominator_ == 1)) {
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// Not initialized, or scale factor is 1.
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return internal_timestamp;
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} else {
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int32_t internal_diff = internal_timestamp - internal_ref_;
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assert(numerator_ > 0); // Should not be possible.
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// Do not update references in this method.
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// Switch |denominator_| and |numerator_| to convert the other way.
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return external_ref_ + (internal_diff * denominator_) / numerator_;
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}
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}
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} // namespace webrtc
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