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665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
86 lines
3.2 KiB
C++
86 lines
3.2 KiB
C++
/*
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* Copyright (c) 2016 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/legacy_encoded_audio_frame.h"
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#include <algorithm>
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#include <memory>
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#include <utility>
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namespace webrtc {
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LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
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rtc::Buffer&& payload)
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: decoder_(decoder), payload_(std::move(payload)) {}
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LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default;
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size_t LegacyEncodedAudioFrame::Duration() const {
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const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
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return (ret < 0) ? 0 : static_cast<size_t>(ret);
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}
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absl::optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
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LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
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AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
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const int ret = decoder_->Decode(
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payload_.data(), payload_.size(), decoder_->SampleRateHz(),
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decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
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if (ret < 0)
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return absl::nullopt;
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return DecodeResult{static_cast<size_t>(ret), speech_type};
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}
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std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
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AudioDecoder* decoder,
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rtc::Buffer&& payload,
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uint32_t timestamp,
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size_t bytes_per_ms,
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uint32_t timestamps_per_ms) {
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RTC_DCHECK(payload.data());
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std::vector<AudioDecoder::ParseResult> results;
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size_t split_size_bytes = payload.size();
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// Find a "chunk size" >= 20 ms and < 40 ms.
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const size_t min_chunk_size = bytes_per_ms * 20;
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if (min_chunk_size >= payload.size()) {
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std::unique_ptr<LegacyEncodedAudioFrame> frame(
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new LegacyEncodedAudioFrame(decoder, std::move(payload)));
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results.emplace_back(timestamp, 0, std::move(frame));
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} else {
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// Reduce the split size by half as long as |split_size_bytes| is at least
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// twice the minimum chunk size (so that the resulting size is at least as
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// large as the minimum chunk size).
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while (split_size_bytes >= 2 * min_chunk_size) {
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split_size_bytes /= 2;
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}
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const uint32_t timestamps_per_chunk = static_cast<uint32_t>(
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split_size_bytes * timestamps_per_ms / bytes_per_ms);
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size_t byte_offset;
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uint32_t timestamp_offset;
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for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
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byte_offset += split_size_bytes,
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timestamp_offset += timestamps_per_chunk) {
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split_size_bytes =
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std::min(split_size_bytes, payload.size() - byte_offset);
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rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes);
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std::unique_ptr<LegacyEncodedAudioFrame> frame(
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new LegacyEncodedAudioFrame(decoder, std::move(new_payload)));
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results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
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}
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}
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return results;
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}
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} // namespace webrtc
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