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/*
* 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/g711/audio_decoder_pcm.h"
#include <utility>
#include "modules/audio_coding/codecs/g711/g711_interface.h"
#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
namespace webrtc {
void AudioDecoderPcmU::Reset() {}
std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
return LegacyEncodedAudioFrame::SplitBySamples(
this, std::move(payload), timestamp, 8 * num_channels_, 8);
}
int AudioDecoderPcmU::SampleRateHz() const {
return 8000;
}
size_t AudioDecoderPcmU::Channels() const {
return num_channels_;
}
int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(SampleRateHz(), 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) *
Channels(); // 1 byte per sample per channel
int16_t temp_type = 1; // Default is speech.
size_t ret =
WebRtcG711_DecodeU(encoded, encoded_len_adjusted, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return static_cast<int>(ret);
}
int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// One encoded byte per sample per channel.
return static_cast<int>(encoded_len / Channels());
}
int AudioDecoderPcmU::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
return PacketDuration(encoded, encoded_len);
}
void AudioDecoderPcmA::Reset() {}
std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
return LegacyEncodedAudioFrame::SplitBySamples(
this, std::move(payload), timestamp, 8 * num_channels_, 8);
}
int AudioDecoderPcmA::SampleRateHz() const {
return 8000;
}
size_t AudioDecoderPcmA::Channels() const {
return num_channels_;
}
int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(SampleRateHz(), 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) *
Channels(); // 1 byte per sample per channel
int16_t temp_type = 1; // Default is speech.
size_t ret =
WebRtcG711_DecodeA(encoded, encoded_len_adjusted, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return static_cast<int>(ret);
}
int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// One encoded byte per sample per channel.
return static_cast<int>(encoded_len / Channels());
}
int AudioDecoderPcmA::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
return PacketDuration(encoded, encoded_len);
}
} // namespace webrtc
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