34efc689c2
This new lib contains the bare minimum to implement an iSAC encoder and decoder. The webrtc files have been copied from the revision as the existing imported files (c8b569e0a7ad0b369e15f0197b3a558699ec8efa).
107 lines
3.5 KiB
C++
107 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/codecs/audio_decoder.h"
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#include <assert.h>
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#include "webrtc/base/checks.h"
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namespace webrtc {
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int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
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int sample_rate_hz, size_t max_decoded_bytes,
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int16_t* decoded, SpeechType* speech_type) {
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int duration = PacketDuration(encoded, encoded_len);
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if (duration >= 0 &&
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duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
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return -1;
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}
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return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
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speech_type);
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}
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int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
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int sample_rate_hz, size_t max_decoded_bytes,
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int16_t* decoded, SpeechType* speech_type) {
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int duration = PacketDurationRedundant(encoded, encoded_len);
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if (duration >= 0 &&
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duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
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return -1;
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}
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return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
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speech_type);
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}
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int AudioDecoder::DecodeInternal(const uint8_t* encoded, size_t encoded_len,
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int sample_rate_hz, int16_t* decoded,
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SpeechType* speech_type) {
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return kNotImplemented;
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}
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int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz, int16_t* decoded,
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SpeechType* speech_type) {
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return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
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speech_type);
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}
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bool AudioDecoder::HasDecodePlc() const { return false; }
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size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
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return 0;
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}
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int AudioDecoder::IncomingPacket(const uint8_t* payload,
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size_t payload_len,
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uint16_t rtp_sequence_number,
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uint32_t rtp_timestamp,
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uint32_t arrival_timestamp) {
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return 0;
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}
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int AudioDecoder::ErrorCode() { return 0; }
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int AudioDecoder::PacketDuration(const uint8_t* encoded,
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size_t encoded_len) const {
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return kNotImplemented;
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}
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int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
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size_t encoded_len) const {
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return kNotImplemented;
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}
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bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
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size_t encoded_len) const {
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return false;
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}
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CNG_dec_inst* AudioDecoder::CngDecoderInstance() {
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FATAL() << "Not a CNG decoder";
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return NULL;
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}
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AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
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switch (type) {
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case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech.
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case 1:
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return kSpeech;
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case 2:
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return kComfortNoise;
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default:
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assert(false);
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return kSpeech;
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}
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}
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} // namespace webrtc
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