blob: c266d38daa831bca01c3accdeac304246e121b9f [file] [log] [blame]
* 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 <memory>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_decoder.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
namespace webrtc {
namespace test {
// Provides an AudioDecoder implementation that delivers audio data from a file.
// The "encoded" input should contain information about what RTP timestamp the
// encoding represents, and how many samples the decoder should produce for that
// encoding. A helper method PrepareEncoded is provided to prepare such
// encodings. If packets are missing, as determined from the timestamps, the
// file reading will skip forward to match the loss.
class FakeDecodeFromFile : public AudioDecoder {
FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input,
int sample_rate_hz,
bool stereo)
: input_(std::move(input)),
stereo_(stereo) {}
~FakeDecodeFromFile() = default;
void Reset() override {}
int SampleRateHz() const override { return sample_rate_hz_; }
size_t Channels() const override { return stereo_ ? 2 : 1; }
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
// Helper method. Writes |timestamp|, |samples| and
// |original_payload_size_bytes| to |encoded| in a format that the
// FakeDecodeFromFile decoder will understand. |encoded| must be at least 12
// bytes long.
static void PrepareEncoded(uint32_t timestamp,
size_t samples,
size_t original_payload_size_bytes,
rtc::ArrayView<uint8_t> encoded);
std::unique_ptr<InputAudioFile> input_;
absl::optional<uint32_t> next_timestamp_from_input_;
const int sample_rate_hz_;
const bool stereo_;
size_t last_decoded_length_ = 0;
bool cng_mode_ = false;
} // namespace test
} // namespace webrtc