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2b742f7eaa
This reverts commit d93b7b91e0.
Reason for revert: Breaks downstream tests
Original change's description:
> Unifying the handling of the events in NetEqInput.
>
> Bug: webrtc:14763
> Change-Id: I9615a9ce41c9b577c4ebd4cdcc9885bfbc5dac48
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293040
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39706}
Bug: webrtc:14763
Change-Id: If076c8fc59a38f011dfa20829f2dd91dd2f914b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299420
Owners-Override: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39711}
75 lines
2.3 KiB
C++
75 lines
2.3 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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~EncodeNetEqInput() override;
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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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absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
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return absl::nullopt;
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}
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std::unique_ptr<PacketData> PopPacket() override;
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void AdvanceOutputEvent() override;
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void AdvanceSetMinimumDelay() override {}
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bool ended() const override;
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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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