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/*
* Copyright (c) 2015 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
#include <memory>
#include <string>
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
namespace test {
// This test class provides a way run NetEQ with an external decoder.
class NetEqExternalDecoderTest {
protected:
static const uint8_t kPayloadType = 95;
static const int kOutputLengthMs = 10;
// The external decoder |decoder| is suppose to be of type |codec|.
NetEqExternalDecoderTest(NetEqDecoder codec,
int sample_rate_hz,
AudioDecoder* decoder);
virtual ~NetEqExternalDecoderTest() { }
// In Init(), we register the external decoder.
void Init();
// Inserts a new packet with |rtp_header| and |payload| of
// |payload_size_bytes| bytes. The |receive_timestamp| is an indication
// of the time when the packet was received, and should be measured with
// the same tick rate as the RTP timestamp of the current payload.
virtual void InsertPacket(WebRtcRTPHeader rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp);
// Get 10 ms of audio data.
void GetOutputAudio(AudioFrame* output);
NetEq* neteq() { return neteq_.get(); }
private:
NetEqDecoder codec_;
std::string name_ = "dummy name";
AudioDecoder* decoder_;
int sample_rate_hz_;
size_t channels_;
std::unique_ptr<NetEq> neteq_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_