Repo created

This commit is contained in:
Fr4nz D13trich 2025-11-22 14:04:28 +01:00
parent 81b91f4139
commit f8c34fa5ee
22732 changed files with 4815320 additions and 2 deletions

View file

@ -0,0 +1,75 @@
/*
* Copyright (c) 2015 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 "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include <utility>
#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "rtc_base/checks.h"
namespace webrtc {
AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels)
: sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) {
RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000)
<< "Unsupported sample rate " << sample_rate_hz;
RTC_DCHECK_GE(num_channels, 1);
}
void AudioDecoderPcm16B::Reset() {}
int AudioDecoderPcm16B::SampleRateHz() const {
return sample_rate_hz_;
}
size_t AudioDecoderPcm16B::Channels() const {
return num_channels_;
}
int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
// Adjust the encoded length down to ensure the same number of samples in each
// channel.
const size_t encoded_len_adjusted =
PacketDuration(encoded, encoded_len) * 2 *
Channels(); // 2 bytes per sample per channel
size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len_adjusted, decoded);
*speech_type = ConvertSpeechType(1);
return static_cast<int>(ret);
}
std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz_, 1000);
return LegacyEncodedAudioFrame::SplitBySamples(
this, std::move(payload), timestamp, samples_per_ms * 2 * num_channels_,
samples_per_ms);
}
int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// Two encoded byte per sample per channel.
return static_cast<int>(encoded_len / (2 * Channels()));
}
int AudioDecoderPcm16B::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
return PacketDuration(encoded, encoded_len);
}
} // namespace webrtc

View file

@ -0,0 +1,54 @@
/*
* Copyright (c) 2015 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
#define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
#include <stddef.h>
#include <stdint.h>
#include <vector>
#include "api/audio_codecs/audio_decoder.h"
#include "rtc_base/buffer.h"
namespace webrtc {
class AudioDecoderPcm16B final : public AudioDecoder {
public:
AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels);
AudioDecoderPcm16B(const AudioDecoderPcm16B&) = delete;
AudioDecoderPcm16B& operator=(const AudioDecoderPcm16B&) = delete;
void Reset() override;
std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
uint32_t timestamp) override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
private:
const int sample_rate_hz_;
const size_t num_channels_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_

View file

@ -0,0 +1,39 @@
/*
* Copyright (c) 2014 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 "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "rtc_base/checks.h"
namespace webrtc {
size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) {
return WebRtcPcm16b_Encode(audio, input_len, encoded);
}
size_t AudioEncoderPcm16B::BytesPerSample() const {
return 2;
}
AudioEncoder::CodecType AudioEncoderPcm16B::GetCodecType() const {
return CodecType::kOther;
}
bool AudioEncoderPcm16B::Config::IsOk() const {
if ((sample_rate_hz != 8000) && (sample_rate_hz != 16000) &&
(sample_rate_hz != 32000) && (sample_rate_hz != 48000))
return false;
return AudioEncoderPcm::Config::IsOk();
}
} // namespace webrtc

View file

@ -0,0 +1,46 @@
/*
* Copyright (c) 2014 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
#define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
namespace webrtc {
class AudioEncoderPcm16B final : public AudioEncoderPcm {
public:
struct Config : public AudioEncoderPcm::Config {
public:
Config() : AudioEncoderPcm::Config(107), sample_rate_hz(8000) {}
bool IsOk() const;
int sample_rate_hz;
};
explicit AudioEncoderPcm16B(const Config& config)
: AudioEncoderPcm(config, config.sample_rate_hz) {}
AudioEncoderPcm16B(const AudioEncoderPcm16B&) = delete;
AudioEncoderPcm16B& operator=(const AudioEncoderPcm16B&) = delete;
protected:
size_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
size_t BytesPerSample() const override;
AudioEncoder::CodecType GetCodecType() const override;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_

View file

@ -0,0 +1,32 @@
/*
* Copyright (c) 2011 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 "modules/audio_coding/codecs/pcm16b/pcm16b.h"
size_t WebRtcPcm16b_Encode(const int16_t* speech,
size_t len,
uint8_t* encoded) {
size_t i;
for (i = 0; i < len; ++i) {
uint16_t s = speech[i];
encoded[2 * i] = s >> 8;
encoded[2 * i + 1] = s;
}
return 2 * len;
}
size_t WebRtcPcm16b_Decode(const uint8_t* encoded,
size_t len,
int16_t* speech) {
size_t i;
for (i = 0; i < len / 2; ++i)
speech[i] = encoded[2 * i] << 8 | encoded[2 * i + 1];
return len / 2;
}

View file

@ -0,0 +1,63 @@
/*
* Copyright (c) 2011 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
#define MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
/*
* Define the fixpoint numeric formats
*/
#include <stddef.h>
#include <stdint.h>
#ifdef __cplusplus
extern "C" {
#endif
/****************************************************************************
* WebRtcPcm16b_Encode(...)
*
* "Encode" a sample vector to 16 bit linear (Encoded standard is big endian)
*
* Input:
* - speech : Input speech vector
* - len : Number of samples in speech vector
*
* Output:
* - encoded : Encoded data vector (big endian 16 bit)
*
* Returned value : Length (in bytes) of coded data.
* Always equal to twice the len input parameter.
*/
size_t WebRtcPcm16b_Encode(const int16_t* speech, size_t len, uint8_t* encoded);
/****************************************************************************
* WebRtcPcm16b_Decode(...)
*
* "Decode" a vector to 16 bit linear (Encoded standard is big endian)
*
* Input:
* - encoded : Encoded data vector (big endian 16 bit)
* - len : Number of bytes in encoded
*
* Output:
* - speech : Decoded speech vector
*
* Returned value : Samples in speech
*/
size_t WebRtcPcm16b_Decode(const uint8_t* encoded, size_t len, int16_t* speech);
#ifdef __cplusplus
}
#endif
#endif /* MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_ */

View file

@ -0,0 +1,29 @@
/*
* Copyright (c) 2017 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 "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include <stdint.h>
#include <initializer_list>
namespace webrtc {
void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs) {
for (uint8_t num_channels : {1, 2}) {
for (int sample_rate_hz : {8000, 16000, 32000}) {
specs->push_back(
{{"L16", sample_rate_hz, num_channels},
{sample_rate_hz, num_channels, sample_rate_hz * num_channels * 16}});
}
}
}
} // namespace webrtc

View file

@ -0,0 +1,22 @@
/*
* Copyright (c) 2017 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
#define MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
#include <vector>
#include "api/audio_codecs/audio_format.h"
namespace webrtc {
void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs);
}
#endif // MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_