aboutsummaryrefslogtreecommitdiff
path: root/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
diff options
context:
space:
mode:
Diffstat (limited to 'webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc')
-rw-r--r--webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc222
1 files changed, 222 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
new file mode 100644
index 0000000000..855a39e675
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
@@ -0,0 +1,222 @@
+/*
+ * 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
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-09 07:04:24 +0000