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c7b4a45594
- AudioFrame - AudioCodingModule BUG=webrtc:4690 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/3019543002 Cr-Original-Commit-Position: refs/heads/master@{#20005} Committed: https://webrtc.googlesource.com/src/+/2d0f77585d556d8b11d6269d35149ae9ca14c472 Review-Url: https://codereview.webrtc.org/3019543002 Cr-Commit-Position: refs/heads/master@{#20019}
242 lines
8.7 KiB
C++
242 lines
8.7 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/acm2/acm_receive_test.h"
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#include <assert.h>
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#include <stdio.h>
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#include <memory>
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "modules/audio_coding/codecs/audio_format_conversion.h"
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_coding/neteq/tools/audio_sink.h"
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "modules/audio_coding/neteq/tools/packet_source.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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namespace {
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// Returns true if the codec should be registered, otherwise false. Changes
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// the number of channels for the Opus codec to always be 1.
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bool ModifyAndUseThisCodec(CodecInst* codec_param) {
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if (STR_CASE_CMP(codec_param->plname, "CN") == 0 &&
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codec_param->plfreq == 48000)
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return false; // Skip 48 kHz comfort noise.
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if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0)
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return false; // Skip DTFM.
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return true;
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}
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// Remaps payload types from ACM's default to those used in the resource file
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// neteq_universal_new.rtp. Returns true if the codec should be registered,
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// otherwise false. The payload types are set as follows (all are mono codecs):
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// PCMu = 0;
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// PCMa = 8;
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// Comfort noise 8 kHz = 13
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// Comfort noise 16 kHz = 98
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// Comfort noise 32 kHz = 99
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// iLBC = 102
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// iSAC wideband = 103
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// iSAC super-wideband = 104
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// AVT/DTMF = 106
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// RED = 117
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// PCM16b 8 kHz = 93
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// PCM16b 16 kHz = 94
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// PCM16b 32 kHz = 95
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// G.722 = 94
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bool RemapPltypeAndUseThisCodec(const char* plname,
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int plfreq,
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size_t channels,
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int* pltype) {
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if (channels != 1)
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return false; // Don't use non-mono codecs.
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// Re-map pltypes to those used in the NetEq test files.
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if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) {
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*pltype = 0;
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} else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) {
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*pltype = 8;
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} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) {
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*pltype = 13;
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} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) {
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*pltype = 98;
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} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) {
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*pltype = 99;
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} else if (STR_CASE_CMP(plname, "ILBC") == 0) {
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*pltype = 102;
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} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) {
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*pltype = 103;
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} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
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*pltype = 104;
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} else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 8000) {
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*pltype = 106;
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} else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 16000) {
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*pltype = 114;
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} else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 32000) {
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*pltype = 115;
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} else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 48000) {
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*pltype = 116;
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} else if (STR_CASE_CMP(plname, "red") == 0) {
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*pltype = 117;
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} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) {
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*pltype = 93;
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} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) {
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*pltype = 94;
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} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
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*pltype = 95;
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} else if (STR_CASE_CMP(plname, "G722") == 0) {
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*pltype = 9;
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} else {
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// Don't use any other codecs.
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return false;
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}
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return true;
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}
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AudioCodingModule::Config MakeAcmConfig(
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Clock* clock,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
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AudioCodingModule::Config config;
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config.clock = clock;
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config.decoder_factory = std::move(decoder_factory);
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return config;
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}
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} // namespace
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AcmReceiveTestOldApi::AcmReceiveTestOldApi(
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PacketSource* packet_source,
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AudioSink* audio_sink,
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int output_freq_hz,
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NumOutputChannels exptected_output_channels,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
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: clock_(0),
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acm_(webrtc::AudioCodingModule::Create(
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MakeAcmConfig(&clock_, std::move(decoder_factory)))),
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packet_source_(packet_source),
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audio_sink_(audio_sink),
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output_freq_hz_(output_freq_hz),
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exptected_output_channels_(exptected_output_channels) {}
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AcmReceiveTestOldApi::~AcmReceiveTestOldApi() = default;
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void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
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CodecInst my_codec_param;
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for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
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ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
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if (ModifyAndUseThisCodec(&my_codec_param)) {
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ASSERT_EQ(true,
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acm_->RegisterReceiveCodec(my_codec_param.pltype,
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CodecInstToSdp(my_codec_param)))
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<< "Couldn't register receive codec.\n";
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}
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}
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}
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void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
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CodecInst my_codec_param;
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for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
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ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
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if (!ModifyAndUseThisCodec(&my_codec_param)) {
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// Skip this codec.
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continue;
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}
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if (RemapPltypeAndUseThisCodec(my_codec_param.plname,
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my_codec_param.plfreq,
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my_codec_param.channels,
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&my_codec_param.pltype)) {
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ASSERT_EQ(true,
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acm_->RegisterReceiveCodec(my_codec_param.pltype,
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CodecInstToSdp(my_codec_param)))
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<< "Couldn't register receive codec.\n";
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}
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}
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}
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void AcmReceiveTestOldApi::Run() {
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for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
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packet = packet_source_->NextPacket()) {
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// Pull audio until time to insert packet.
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while (clock_.TimeInMilliseconds() < packet->time_ms()) {
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AudioFrame output_frame;
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bool muted;
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EXPECT_EQ(0,
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acm_->PlayoutData10Ms(output_freq_hz_, &output_frame, &muted));
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ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
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ASSERT_FALSE(muted);
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const size_t samples_per_block =
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static_cast<size_t>(output_freq_hz_ * 10 / 1000);
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EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
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if (exptected_output_channels_ != kArbitraryChannels) {
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if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
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// Don't check number of channels for PLC output, since each test run
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// usually starts with a short period of mono PLC before decoding the
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// first packet.
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} else {
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EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
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}
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}
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ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
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clock_.AdvanceTimeMilliseconds(10);
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AfterGetAudio();
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}
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// Insert packet after converting from RTPHeader to WebRtcRTPHeader.
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WebRtcRTPHeader header;
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header.header = packet->header();
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header.frameType = kAudioFrameSpeech;
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memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
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EXPECT_EQ(0,
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acm_->IncomingPacket(
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packet->payload(),
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static_cast<int32_t>(packet->payload_length_bytes()),
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header))
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<< "Failure when inserting packet:" << std::endl
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<< " PT = " << static_cast<int>(header.header.payloadType) << std::endl
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<< " TS = " << header.header.timestamp << std::endl
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<< " SN = " << header.header.sequenceNumber;
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}
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}
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AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
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PacketSource* packet_source,
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AudioSink* audio_sink,
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int output_freq_hz_1,
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int output_freq_hz_2,
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int toggle_period_ms,
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NumOutputChannels exptected_output_channels)
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: AcmReceiveTestOldApi(packet_source,
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audio_sink,
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output_freq_hz_1,
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exptected_output_channels,
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CreateBuiltinAudioDecoderFactory()),
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output_freq_hz_1_(output_freq_hz_1),
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output_freq_hz_2_(output_freq_hz_2),
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toggle_period_ms_(toggle_period_ms),
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last_toggle_time_ms_(clock_.TimeInMilliseconds()) {}
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void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
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if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
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output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
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? output_freq_hz_2_
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: output_freq_hz_1_;
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last_toggle_time_ms_ = clock_.TimeInMilliseconds();
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}
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}
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} // namespace test
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} // namespace webrtc
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