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/*
* 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 "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
#include <assert.h>
#include <stdio.h>
#include <string.h>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
namespace webrtc {
namespace test {
AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
int source_rate_hz,
int test_duration_ms)
: clock_(0),
acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
audio_source_(audio_source),
source_rate_hz_(source_rate_hz),
input_block_size_samples_(
static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),
codec_registered_(false),
test_duration_ms_(test_duration_ms),
frame_type_(kAudioFrameSpeech),
payload_type_(0),
timestamp_(0),
sequence_number_(0) {
input_frame_.sample_rate_hz_ = source_rate_hz_;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = input_block_size_samples_;
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
acm_->RegisterTransportCallback(this);
}
bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
int sampling_freq_hz,
int channels,
int payload_type,
int frame_size_samples) {
CodecInst codec;
RTC_CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec,
sampling_freq_hz, channels));
codec.pltype = payload_type;
codec.pacsize = frame_size_samples;
codec_registered_ = (acm_->RegisterSendCodec(codec) == 0);
input_frame_.num_channels_ = channels;
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
return codec_registered_;
}
bool AcmSendTestOldApi::RegisterExternalCodec(
AudioEncoder* external_speech_encoder) {
acm_->RegisterExternalSendCodec(external_speech_encoder);
input_frame_.num_channels_ = external_speech_encoder->NumChannels();
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
return codec_registered_ = true;
}
Packet* AcmSendTestOldApi::NextPacket() {
assert(codec_registered_);
if (filter_.test(static_cast<size_t>(payload_type_))) {
// This payload type should be filtered out. Since the payload type is the
// same throughout the whole test run, no packet at all will be delivered.
// We can just as well signal that the test is over by returning NULL.
return NULL;
}
// Insert audio and process until one packet is produced.
while (clock_.TimeInMilliseconds() < test_duration_ms_) {
clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
RTC_CHECK(
audio_source_->Read(input_block_size_samples_, input_frame_.data_));
if (input_frame_.num_channels_ > 1) {
InputAudioFile::DuplicateInterleaved(input_frame_.data_,
input_block_size_samples_,
input_frame_.num_channels_,
input_frame_.data_);
}
data_to_send_ = false;
RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
if (data_to_send_) {
// Encoded packet received.
return CreatePacket();
}
}
// Test ended.
return NULL;
}
// This method receives the callback from ACM when a new packet is produced.
int32_t AcmSendTestOldApi::SendData(
FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) {
// Store the packet locally.
frame_type_ = frame_type;
payload_type_ = payload_type;
timestamp_ = timestamp;
last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
assert(last_payload_vec_.size() == payload_len_bytes);
data_to_send_ = true;
return 0;
}
Packet* AcmSendTestOldApi::CreatePacket() {
const size_t kRtpHeaderSize = 12;
size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize;
uint8_t* packet_memory = new uint8_t[allocated_bytes];
// Populate the header bytes.
packet_memory[0] = 0x80;
packet_memory[1] = static_cast<uint8_t>(payload_type_);
packet_memory[2] = (sequence_number_ >> 8) & 0xFF;
packet_memory[3] = (sequence_number_) & 0xFF;
packet_memory[4] = (timestamp_ >> 24) & 0xFF;
packet_memory[5] = (timestamp_ >> 16) & 0xFF;
packet_memory[6] = (timestamp_ >> 8) & 0xFF;
packet_memory[7] = timestamp_ & 0xFF;
// Set SSRC to 0x12345678.
packet_memory[8] = 0x12;
packet_memory[9] = 0x34;
packet_memory[10] = 0x56;
packet_memory[11] = 0x78;
++sequence_number_;
// Copy the payload data.
memcpy(packet_memory + kRtpHeaderSize,
&last_payload_vec_[0],
last_payload_vec_.size());
Packet* packet =
new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds());
assert(packet);
assert(packet->valid_header());
return packet;
}
} // namespace test
} // namespace webrtc
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