blob: 4c50a4b5551c78964d3b7187dc7a82bfb2d18049 [file] [log] [blame]
* Copyright (c) 2012 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 <math.h>
#include <memory>
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
namespace webrtc {
enum StereoMonoMode { kNotSet, kMono, kStereo };
class TestPackStereo : public AudioPacketizationCallback {
void RegisterReceiverACM(AudioCodingModule* acm);
int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_size,
int64_t absolute_capture_timestamp_ms) override;
uint16_t payload_size();
uint32_t timestamp_diff();
void reset_payload_size();
void set_codec_mode(StereoMonoMode mode);
void set_lost_packet(bool lost);
AudioCodingModule* receiver_acm_;
int16_t seq_no_;
uint32_t timestamp_diff_;
uint32_t last_in_timestamp_;
uint64_t total_bytes_;
int payload_size_;
StereoMonoMode codec_mode_;
// Simulate packet losses
bool lost_packet_;
class TestStereo {
void Perform();
// The default value of '-1' indicates that the registration is based only on
// codec name and a sampling frequncy matching is not required. This is useful
// for codecs which support several sampling frequency.
void RegisterSendCodec(char side,
char* codec_name,
int32_t samp_freq_hz,
int rate,
int pack_size,
int channels);
void Run(TestPackStereo* channel,
int in_channels,
int out_channels,
int percent_loss = 0);
void OpenOutFile(int16_t test_number);
std::unique_ptr<AudioCodingModule> acm_a_;
std::unique_ptr<AudioCodingModule> acm_b_;
TestPackStereo* channel_a2b_;
PCMFile* in_file_stereo_;
PCMFile* in_file_mono_;
PCMFile out_file_;
int16_t test_cntr_;
uint16_t pack_size_samp_;
uint16_t pack_size_bytes_;
int counter_;
char* send_codec_name_;
} // namespace webrtc