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Unified Diff: webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc

Issue 2326003002: Moved codec-specific audio packet splitting into decoders. (Closed)
Patch Set: Fixed types in packet splitting (size_t vs. uint32_t) Created 4 years, 3 months ago
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Index: webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
diff --git a/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
new file mode 100644
index 0000000000000000000000000000000000000000..5e6ff01f2227bc601d72e093cbb197f96d468e52
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
@@ -0,0 +1,105 @@
+/*
+ * 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 "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+namespace webrtc {
+
+LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
+ rtc::Buffer&& payload,
+ bool is_primary_payload)
+ : decoder_(decoder),
+ payload_(std::move(payload)),
+ is_primary_payload_(is_primary_payload) {}
+
+LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default;
+
+size_t LegacyEncodedAudioFrame::Duration() const {
+ int ret;
+ if (is_primary_payload_) {
+ ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+ } else {
+ ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
+ }
+ return (ret < 0) ? 0 : static_cast<size_t>(ret);
+}
+
+rtc::Optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
+LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
+ AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+ int ret;
+ if (is_primary_payload_) {
+ ret = decoder_->Decode(
+ payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+ decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+ } else {
+ ret = decoder_->DecodeRedundant(
+ payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+ decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+ }
+
+ if (ret < 0)
+ return rtc::Optional<DecodeResult>();
+
+ return rtc::Optional<DecodeResult>({static_cast<size_t>(ret), speech_type});
+}
+
+std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
+ AudioDecoder* decoder,
+ rtc::Buffer&& payload,
+ uint32_t timestamp,
+ bool is_primary,
+ 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), is_primary));
+ results.emplace_back(timestamp, is_primary, 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),
+ is_primary));
+ results.emplace_back(timestamp + timestamp_offset, is_primary,
+ std::move(frame));
+ }
+ }
+
+ return results;
+}
+
+} // namespace webrtc

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