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https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
64 lines
1.8 KiB
C++
64 lines
1.8 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/main/acm2/acm_pcmu.h"
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#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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// Codec interface.
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namespace webrtc {
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namespace acm2 {
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ACMPCMU::ACMPCMU(int16_t codec_id) { codec_id_ = codec_id; }
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ACMPCMU::~ACMPCMU() {}
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int16_t ACMPCMU::InternalEncode(uint8_t* bitstream,
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int16_t* bitstream_len_byte) {
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*bitstream_len_byte = WebRtcG711_EncodeU(
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NULL, &in_audio_[in_audio_ix_read_], frame_len_smpl_ * num_channels_,
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reinterpret_cast<int16_t*>(bitstream));
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// Increment the read index this tell the caller that how far
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// we have gone forward in reading the audio buffer.
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in_audio_ix_read_ += frame_len_smpl_ * num_channels_;
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return *bitstream_len_byte;
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}
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int16_t ACMPCMU::InternalInitEncoder(WebRtcACMCodecParams* /* codec_params */) {
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// This codec does not need initialization, PCM has no instance.
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return 0;
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}
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ACMGenericCodec* ACMPCMU::CreateInstance(void) { return NULL; }
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int16_t ACMPCMU::InternalCreateEncoder() {
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// PCM has no instance.
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return 0;
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}
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void ACMPCMU::InternalDestructEncoderInst(void* /* ptr_inst */) {
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// PCM has no instance.
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}
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void ACMPCMU::DestructEncoderSafe() {
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// PCM has no instance.
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encoder_exist_ = false;
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encoder_initialized_ = false;
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}
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} // namespace acm2
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} // namespace webrtc
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