blob: ce68196a3f474a6777ced8cd3a91316b83e1a141 [file] [log] [blame]
* Copyright (c) 2014 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 <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
class AudioEncoder;
namespace test {
class InputAudioFile;
class Packet;
class AcmSendTestOldApi : public AudioPacketizationCallback,
public PacketSource {
AcmSendTestOldApi(InputAudioFile* audio_source,
int source_rate_hz,
int test_duration_ms);
virtual ~AcmSendTestOldApi() {}
// Registers the send codec. Returns true on success, false otherwise.
bool RegisterCodec(const char* payload_name,
int sampling_freq_hz,
int channels,
int payload_type,
int frame_size_samples);
// Registers an external send codec. Returns true on success, false otherwise.
bool RegisterExternalCodec(AudioEncoder* external_speech_encoder);
// Returns the next encoded packet. Returns NULL if the test duration was
// exceeded. Ownership of the packet is handed over to the caller.
// Inherited from PacketSource.
Packet* NextPacket();
// Inherited from AudioPacketizationCallback.
int32_t SendData(FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) override;
AudioCodingModule* acm() { return acm_.get(); }
static const int kBlockSizeMs = 10;
// Creates a Packet object from the last packet produced by ACM (and received
// through the SendData method as a callback). Ownership of the new Packet
// object is transferred to the caller.
Packet* CreatePacket();
SimulatedClock clock_;
rtc::scoped_ptr<AudioCodingModule> acm_;
InputAudioFile* audio_source_;
int source_rate_hz_;
const size_t input_block_size_samples_;
AudioFrame input_frame_;
bool codec_registered_;
int test_duration_ms_;
// The following member variables are set whenever SendData() is called.
FrameType frame_type_;
int payload_type_;
uint32_t timestamp_;
uint16_t sequence_number_;
std::vector<uint8_t> last_payload_vec_;
bool data_to_send_;
} // namespace test
} // namespace webrtc