dce40cf804
use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
107 lines
3.5 KiB
C++
107 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/codecs/audio_decoder.h"
|
|
|
|
#include <assert.h>
|
|
|
|
#include "webrtc/base/checks.h"
|
|
|
|
namespace webrtc {
|
|
|
|
int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
|
|
int sample_rate_hz, size_t max_decoded_bytes,
|
|
int16_t* decoded, SpeechType* speech_type) {
|
|
int duration = PacketDuration(encoded, encoded_len);
|
|
if (duration >= 0 &&
|
|
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
|
|
return -1;
|
|
}
|
|
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
|
|
speech_type);
|
|
}
|
|
|
|
int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
|
|
int sample_rate_hz, size_t max_decoded_bytes,
|
|
int16_t* decoded, SpeechType* speech_type) {
|
|
int duration = PacketDurationRedundant(encoded, encoded_len);
|
|
if (duration >= 0 &&
|
|
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
|
|
return -1;
|
|
}
|
|
return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
|
|
speech_type);
|
|
}
|
|
|
|
int AudioDecoder::DecodeInternal(const uint8_t* encoded, size_t encoded_len,
|
|
int sample_rate_hz, int16_t* decoded,
|
|
SpeechType* speech_type) {
|
|
return kNotImplemented;
|
|
}
|
|
|
|
int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
|
|
size_t encoded_len,
|
|
int sample_rate_hz, int16_t* decoded,
|
|
SpeechType* speech_type) {
|
|
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
|
|
speech_type);
|
|
}
|
|
|
|
bool AudioDecoder::HasDecodePlc() const { return false; }
|
|
|
|
size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
|
|
return 0;
|
|
}
|
|
|
|
int AudioDecoder::IncomingPacket(const uint8_t* payload,
|
|
size_t payload_len,
|
|
uint16_t rtp_sequence_number,
|
|
uint32_t rtp_timestamp,
|
|
uint32_t arrival_timestamp) {
|
|
return 0;
|
|
}
|
|
|
|
int AudioDecoder::ErrorCode() { return 0; }
|
|
|
|
int AudioDecoder::PacketDuration(const uint8_t* encoded,
|
|
size_t encoded_len) const {
|
|
return kNotImplemented;
|
|
}
|
|
|
|
int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
|
|
size_t encoded_len) const {
|
|
return kNotImplemented;
|
|
}
|
|
|
|
bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
|
|
size_t encoded_len) const {
|
|
return false;
|
|
}
|
|
|
|
CNG_dec_inst* AudioDecoder::CngDecoderInstance() {
|
|
FATAL() << "Not a CNG decoder";
|
|
return NULL;
|
|
}
|
|
|
|
AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
|
|
switch (type) {
|
|
case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech.
|
|
case 1:
|
|
return kSpeech;
|
|
case 2:
|
|
return kComfortNoise;
|
|
default:
|
|
assert(false);
|
|
return kSpeech;
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|