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c4ed5f0b1a
Bug: chromium:1279775 Change-Id: I8cc3f5fe25b9e707e9d171251026bd5a8bad5da5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251844 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36036}
102 lines
3.5 KiB
C++
102 lines
3.5 KiB
C++
/*
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* Copyright (c) 2015 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 "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
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#include <utility>
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#include "modules/audio_coding/codecs/g711/g711_interface.h"
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#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
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namespace webrtc {
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void AudioDecoderPcmU::Reset() {}
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std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload(
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rtc::Buffer&& payload,
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uint32_t timestamp) {
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return LegacyEncodedAudioFrame::SplitBySamples(
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this, std::move(payload), timestamp, 8 * num_channels_, 8);
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}
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int AudioDecoderPcmU::SampleRateHz() const {
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return 8000;
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}
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size_t AudioDecoderPcmU::Channels() const {
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return num_channels_;
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}
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int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) {
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RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
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// Adjust the encoded length down to ensure the same number of samples in each
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// channel.
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const size_t encoded_len_adjusted =
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PacketDuration(encoded, encoded_len) *
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Channels(); // 1 byte per sample per channel
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int16_t temp_type = 1; // Default is speech.
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size_t ret =
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WebRtcG711_DecodeU(encoded, encoded_len_adjusted, decoded, &temp_type);
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*speech_type = ConvertSpeechType(temp_type);
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return static_cast<int>(ret);
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}
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int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
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size_t encoded_len) const {
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// One encoded byte per sample per channel.
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return static_cast<int>(encoded_len / Channels());
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}
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void AudioDecoderPcmA::Reset() {}
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std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
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rtc::Buffer&& payload,
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uint32_t timestamp) {
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return LegacyEncodedAudioFrame::SplitBySamples(
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this, std::move(payload), timestamp, 8 * num_channels_, 8);
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}
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int AudioDecoderPcmA::SampleRateHz() const {
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return 8000;
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}
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size_t AudioDecoderPcmA::Channels() const {
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return num_channels_;
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}
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int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) {
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RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
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// Adjust the encoded length down to ensure the same number of samples in each
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// channel.
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const size_t encoded_len_adjusted =
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PacketDuration(encoded, encoded_len) *
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Channels(); // 1 byte per sample per channel
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int16_t temp_type = 1; // Default is speech.
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size_t ret =
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WebRtcG711_DecodeA(encoded, encoded_len_adjusted, decoded, &temp_type);
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*speech_type = ConvertSpeechType(temp_type);
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return static_cast<int>(ret);
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}
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int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
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size_t encoded_len) const {
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// One encoded byte per sample per channel.
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return static_cast<int>(encoded_len / Channels());
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}
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} // namespace webrtc
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