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/*
* Copyright (c) 2016 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/legacy_encoded_audio_frame.h"
#include <algorithm>
#include <memory>
#include <utility>
#include "rtc_base/checks.h"
namespace webrtc {
LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
rtc::Buffer&& payload)
: decoder_(decoder), payload_(std::move(payload)) {}
LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default;
size_t LegacyEncodedAudioFrame::Duration() const {
const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
return (ret < 0) ? 0 : static_cast<size_t>(ret);
}
std::optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
const int ret = decoder_->Decode(
payload_.data(), payload_.size(), decoder_->SampleRateHz(),
decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
if (ret < 0)
return std::nullopt;
return DecodeResult{static_cast<size_t>(ret), speech_type};
}
std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
AudioDecoder* decoder,
rtc::Buffer&& payload,
uint32_t timestamp,
size_t bytes_per_ms,
uint32_t timestamps_per_ms) {
RTC_DCHECK(payload.data());
std::vector<AudioDecoder::ParseResult> results;
size_t split_size_bytes = payload.size();
// Find a "chunk size" >= 20 ms and < 40 ms.
const size_t min_chunk_size = bytes_per_ms * 20;
if (min_chunk_size >= payload.size()) {
std::unique_ptr<LegacyEncodedAudioFrame> frame(
new LegacyEncodedAudioFrame(decoder, std::move(payload)));
results.emplace_back(timestamp, 0, std::move(frame));
} else {
// Reduce the split size by half as long as `split_size_bytes` is at least
// twice the minimum chunk size (so that the resulting size is at least as
// large as the minimum chunk size).
while (split_size_bytes >= 2 * min_chunk_size) {
split_size_bytes /= 2;
}
const uint32_t timestamps_per_chunk = static_cast<uint32_t>(
split_size_bytes * timestamps_per_ms / bytes_per_ms);
size_t byte_offset;
uint32_t timestamp_offset;
for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
byte_offset += split_size_bytes,
timestamp_offset += timestamps_per_chunk) {
split_size_bytes =
std::min(split_size_bytes, payload.size() - byte_offset);
rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes);
std::unique_ptr<LegacyEncodedAudioFrame> frame(
new LegacyEncodedAudioFrame(decoder, std::move(new_payload)));
results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
}
}
return results;
}
} // namespace webrtc
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