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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}
73 lines
2.6 KiB
C++
73 lines
2.6 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_FAKE_DECODE_FROM_FILE_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
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#include <memory>
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#include "api/array_view.h"
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#include "api/audio_codecs/audio_decoder.h"
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#include "api/optional.h"
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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namespace webrtc {
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namespace test {
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// Provides an AudioDecoder implementation that delivers audio data from a file.
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// The "encoded" input should contain information about what RTP timestamp the
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// encoding represents, and how many samples the decoder should produce for that
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// encoding. A helper method PrepareEncoded is provided to prepare such
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// encodings. If packets are missing, as determined from the timestamps, the
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// file reading will skip forward to match the loss.
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class FakeDecodeFromFile : public AudioDecoder {
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public:
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FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input,
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int sample_rate_hz,
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bool stereo)
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: input_(std::move(input)),
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sample_rate_hz_(sample_rate_hz),
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stereo_(stereo) {}
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~FakeDecodeFromFile() = default;
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void Reset() override {}
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int SampleRateHz() const override { return sample_rate_hz_; }
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size_t Channels() const override { return stereo_ ? 2 : 1; }
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int DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) override;
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// Helper method. Writes |timestamp|, |samples| and
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// |original_payload_size_bytes| to |encoded| in a format that the
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// FakeDecodeFromFile decoder will understand. |encoded| must be at least 12
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// bytes long.
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static void PrepareEncoded(uint32_t timestamp,
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size_t samples,
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size_t original_payload_size_bytes,
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rtc::ArrayView<uint8_t> encoded);
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private:
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std::unique_ptr<InputAudioFile> input_;
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rtc::Optional<uint32_t> next_timestamp_from_input_;
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const int sample_rate_hz_;
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const bool stereo_;
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size_t last_decoded_length_ = 0;
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bool cng_mode_ = false;
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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_FAKE_DECODE_FROM_FILE_H_
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