Index: webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc |
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc |
deleted file mode 100644 |
index ac38dc011d74227e8dec0b1997367ee69f8f2efb..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc |
+++ /dev/null |
@@ -1,158 +0,0 @@ |
-/* |
- * 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 |