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b602123a5a
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
70 lines
2.2 KiB
C++
70 lines
2.2 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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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
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#include <memory>
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#include "api/audio_codecs/audio_encoder.h"
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#include "modules/audio_coding/neteq/tools/neteq_input.h"
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namespace webrtc {
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namespace test {
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// This class provides a NetEqInput that takes audio from a generator object and
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// encodes it using a given audio encoder.
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class EncodeNetEqInput : public NetEqInput {
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public:
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// Generator class, to be provided to the EncodeNetEqInput constructor.
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class Generator {
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public:
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virtual ~Generator() = default;
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// Returns the next num_samples values from the signal generator.
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virtual rtc::ArrayView<const int16_t> Generate(size_t num_samples) = 0;
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};
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// The source will end after the given input duration.
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EncodeNetEqInput(std::unique_ptr<Generator> generator,
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std::unique_ptr<AudioEncoder> encoder,
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int64_t input_duration_ms);
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absl::optional<int64_t> NextPacketTime() const override;
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absl::optional<int64_t> NextOutputEventTime() const override;
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std::unique_ptr<PacketData> PopPacket() override;
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void AdvanceOutputEvent() override;
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bool ended() const override {
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return next_output_event_ms_ <= input_duration_ms_;
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}
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absl::optional<RTPHeader> NextHeader() const override;
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private:
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static constexpr int64_t kOutputPeriodMs = 10;
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void CreatePacket();
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std::unique_ptr<Generator> generator_;
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std::unique_ptr<AudioEncoder> encoder_;
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std::unique_ptr<PacketData> packet_data_;
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uint32_t rtp_timestamp_ = 0;
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int16_t sequence_number_ = 0;
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int64_t next_packet_time_ms_ = 0;
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int64_t next_output_event_ms_ = 0;
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const int64_t input_duration_ms_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
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