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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
71 lines
2.2 KiB
C++
71 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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#include "modules/include/module_common_types.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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rtc::Optional<int64_t> NextPacketTime() const override;
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rtc::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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rtc::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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