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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/acm2/acm_receive_test_oldapi.h"
#include <assert.h>
#include <stdio.h>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
namespace webrtc {
namespace test {
namespace {
// Returns true if the codec should be registered, otherwise false. Changes
// the number of channels for the Opus codec to always be 1.
bool ModifyAndUseThisCodec(CodecInst* codec_param) {
if (STR_CASE_CMP(codec_param->plname, "CN") == 0 &&
codec_param->plfreq == 48000)
return false; // Skip 48 kHz comfort noise.
if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0)
return false; // Skip DTFM.
return true;
}
// Remaps payload types from ACM's default to those used in the resource file
// neteq_universal_new.rtp. Returns true if the codec should be registered,
// otherwise false. The payload types are set as follows (all are mono codecs):
// PCMu = 0;
// PCMa = 8;
// Comfort noise 8 kHz = 13
// Comfort noise 16 kHz = 98
// Comfort noise 32 kHz = 99
// iLBC = 102
// iSAC wideband = 103
// iSAC super-wideband = 104
// AVT/DTMF = 106
// RED = 117
// PCM16b 8 kHz = 93
// PCM16b 16 kHz = 94
// PCM16b 32 kHz = 95
// G.722 = 94
bool RemapPltypeAndUseThisCodec(const char* plname,
int plfreq,
size_t channels,
int* pltype) {
if (channels != 1)
return false; // Don't use non-mono codecs.
// Re-map pltypes to those used in the NetEq test files.
if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) {
*pltype = 0;
} else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) {
*pltype = 8;
} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) {
*pltype = 13;
} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) {
*pltype = 98;
} else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) {
*pltype = 99;
} else if (STR_CASE_CMP(plname, "ILBC") == 0) {
*pltype = 102;
} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) {
*pltype = 103;
} else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
*pltype = 104;
} else if (STR_CASE_CMP(plname, "telephone-event") == 0) {
*pltype = 106;
} else if (STR_CASE_CMP(plname, "red") == 0) {
*pltype = 117;
} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) {
*pltype = 93;
} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) {
*pltype = 94;
} else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
*pltype = 95;
} else if (STR_CASE_CMP(plname, "G722") == 0) {
*pltype = 9;
} else {
// Don't use any other codecs.
return false;
}
return true;
}
} // namespace
AcmReceiveTestOldApi::AcmReceiveTestOldApi(
PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz,
NumOutputChannels exptected_output_channels)
: clock_(0),
acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
packet_source_(packet_source),
audio_sink_(audio_sink),
output_freq_hz_(output_freq_hz),
exptected_output_channels_(exptected_output_channels) {
}
void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
CodecInst my_codec_param;
for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
if (ModifyAndUseThisCodec(&my_codec_param)) {
ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
<< "Couldn't register receive codec.\n";
}
}
}
void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
CodecInst my_codec_param;
for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
if (!ModifyAndUseThisCodec(&my_codec_param)) {
// Skip this codec.
continue;
}
if (RemapPltypeAndUseThisCodec(my_codec_param.plname,
my_codec_param.plfreq,
my_codec_param.channels,
&my_codec_param.pltype)) {
ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
<< "Couldn't register receive codec.\n";
}
}
}
int AcmReceiveTestOldApi::RegisterExternalReceiveCodec(
int rtp_payload_type,
AudioDecoder* external_decoder,
int sample_rate_hz,
int num_channels,
const std::string& name) {
return acm_->RegisterExternalReceiveCodec(rtp_payload_type, external_decoder,
sample_rate_hz, num_channels, name);
}
void AcmReceiveTestOldApi::Run() {
for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
packet.reset(packet_source_->NextPacket())) {
// Pull audio until time to insert packet.
while (clock_.TimeInMilliseconds() < packet->time_ms()) {
AudioFrame output_frame;
EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
const size_t samples_per_block =
static_cast<size_t>(output_freq_hz_ * 10 / 1000);
EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
if (exptected_output_channels_ != kArbitraryChannels) {
if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
// Don't check number of channels for PLC output, since each test run
// usually starts with a short period of mono PLC before decoding the
// first packet.
} else {
EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
}
}
ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
clock_.AdvanceTimeMilliseconds(10);
AfterGetAudio();
}
// Insert packet after converting from RTPHeader to WebRtcRTPHeader.
WebRtcRTPHeader header;
header.header = packet->header();
header.frameType = kAudioFrameSpeech;
memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
EXPECT_EQ(0,
acm_->IncomingPacket(
packet->payload(),
static_cast<int32_t>(packet->payload_length_bytes()),
header))
<< "Failure when inserting packet:" << std::endl
<< " PT = " << static_cast<int>(header.header.payloadType) << std::endl
<< " TS = " << header.header.timestamp << std::endl
<< " SN = " << header.header.sequenceNumber;
}
}
AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz_1,
int output_freq_hz_2,
int toggle_period_ms,
NumOutputChannels exptected_output_channels)
: AcmReceiveTestOldApi(packet_source,
audio_sink,
output_freq_hz_1,
exptected_output_channels),
output_freq_hz_1_(output_freq_hz_1),
output_freq_hz_2_(output_freq_hz_2),
toggle_period_ms_(toggle_period_ms),
last_toggle_time_ms_(clock_.TimeInMilliseconds()) {
}
void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
? output_freq_hz_2_
: output_freq_hz_1_;
last_toggle_time_ms_ = clock_.TimeInMilliseconds();
}
}
} // namespace test
} // namespace webrtc
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