diff options
Diffstat (limited to 'webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc')
| -rw-r--r-- | webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc | 158 |
1 files changed, 158 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc new file mode 100644 index 0000000000..3a89a77487 --- /dev/null +++ b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc @@ -0,0 +1,158 @@ +/* + * 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/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/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 |