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9b7f3649af
Bug: chromium:1521761 Change-Id: Id5292e80fd6ecae2c39a446dec010b0383bd805e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337200 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41645}
75 lines
2.7 KiB
C++
75 lines
2.7 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/pcm16b/audio_decoder_pcm16b.h"
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#include <utility>
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#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
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#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels)
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: sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) {
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RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
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sample_rate_hz == 32000 || sample_rate_hz == 48000)
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<< "Unsupported sample rate " << sample_rate_hz;
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RTC_DCHECK_GE(num_channels, 1);
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}
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void AudioDecoderPcm16B::Reset() {}
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int AudioDecoderPcm16B::SampleRateHz() const {
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return sample_rate_hz_;
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}
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size_t AudioDecoderPcm16B::Channels() const {
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return num_channels_;
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}
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int AudioDecoderPcm16B::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(sample_rate_hz_, 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) * 2 *
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Channels(); // 2 bytes per sample per channel
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size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len_adjusted, decoded);
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*speech_type = ConvertSpeechType(1);
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return static_cast<int>(ret);
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}
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std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
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rtc::Buffer&& payload,
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uint32_t timestamp) {
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const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz_, 1000);
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return LegacyEncodedAudioFrame::SplitBySamples(
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this, std::move(payload), timestamp, samples_per_ms * 2 * num_channels_,
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samples_per_ms);
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}
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int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
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size_t encoded_len) const {
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// Two encoded byte per sample per channel.
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return static_cast<int>(encoded_len / (2 * Channels()));
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}
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int AudioDecoderPcm16B::PacketDurationRedundant(const uint8_t* encoded,
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size_t encoded_len) const {
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return PacketDuration(encoded, encoded_len);
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}
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} // namespace webrtc
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