Reland "Remove CodecInst pt.1"
This is a reland of 056f9738bf7a3d16da45398239656e165c4e0851 Original change's description: > Remove CodecInst pt.1 > > Update audio_coding tests to not use CodecInst. > > Bug: webrtc:7626 > Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2 > Reviewed-on: https://webrtc-review.googlesource.com/c/112594 > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25879} Bug: webrtc:7626 Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90 Reviewed-on: https://webrtc-review.googlesource.com/c/112942 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25902}
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Fredrik Solenberg
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657b296ff5
@ -14,13 +14,11 @@
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#include <limits>
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#include <string>
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#include "absl/strings/match.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include "common_types.h" // NOLINT(build/include)
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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/include/audio_coding_module_typedefs.h"
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#include "modules/audio_coding/test/utility.h"
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#include "modules/include/module_common_types.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/stringencode.h"
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#include "rtc_base/strings/string_builder.h"
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@ -35,6 +33,11 @@
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// The test loops through all available mono codecs, encode at "a" sends over
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// the channel, and decodes at "b".
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#define CHECK_ERROR(f) \
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do { \
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EXPECT_GE(f, 0) << "Error Calling API"; \
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} while (0)
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namespace {
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const size_t kVariableSize = std::numeric_limits<size_t>::max();
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}
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@ -101,7 +104,7 @@ void TestPack::reset_payload_size() {
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payload_size_ = 0;
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}
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TestAllCodecs::TestAllCodecs(int test_mode)
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TestAllCodecs::TestAllCodecs()
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: acm_a_(AudioCodingModule::Create(
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AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
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acm_b_(AudioCodingModule::Create(
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@ -110,8 +113,6 @@ TestAllCodecs::TestAllCodecs(int test_mode)
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test_count_(0),
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packet_size_samples_(0),
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packet_size_bytes_(0) {
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// test_mode = 0 for silent test (auto test)
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test_mode_ = test_mode;
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}
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TestAllCodecs::~TestAllCodecs() {
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@ -126,23 +127,28 @@ void TestAllCodecs::Perform() {
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webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
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infile_a_.Open(file_name, 32000, "rb");
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if (test_mode_ == 0) {
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RTC_LOG(LS_INFO) << "---------- TestAllCodecs ----------";
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}
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acm_a_->InitializeReceiver();
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acm_b_->InitializeReceiver();
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uint8_t num_encoders = acm_a_->NumberOfCodecs();
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CodecInst my_codec_param;
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for (uint8_t n = 0; n < num_encoders; n++) {
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acm_b_->Codec(n, &my_codec_param);
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if (!strcmp(my_codec_param.plname, "opus")) {
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my_codec_param.channels = 1;
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}
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acm_b_->RegisterReceiveCodec(my_codec_param.pltype,
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CodecInstToSdp(my_codec_param));
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}
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acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
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{104, {"ISAC", 32000, 1}},
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{107, {"L16", 8000, 1}},
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{108, {"L16", 16000, 1}},
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{109, {"L16", 32000, 1}},
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{111, {"L16", 8000, 2}},
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{112, {"L16", 16000, 2}},
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{113, {"L16", 32000, 2}},
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{0, {"PCMU", 8000, 1}},
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{110, {"PCMU", 8000, 2}},
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{8, {"PCMA", 8000, 1}},
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{118, {"PCMA", 8000, 2}},
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{102, {"ILBC", 8000, 1}},
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{9, {"G722", 8000, 1}},
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{119, {"G722", 8000, 2}},
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{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
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{13, {"CN", 8000, 1}},
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{98, {"CN", 16000, 1}},
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{99, {"CN", 32000, 1}}});
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// Create and connect the channel
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channel_a_to_b_ = new TestPack;
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@ -151,9 +157,6 @@ void TestAllCodecs::Perform() {
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// All codecs are tested for all allowed sampling frequencies, rates and
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// packet sizes.
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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test_count_++;
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OpenOutFile(test_count_);
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char codec_g722[] = "G722";
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@ -171,9 +174,6 @@ void TestAllCodecs::Perform() {
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Run(channel_a_to_b_);
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outfile_b_.Close();
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#ifdef WEBRTC_CODEC_ILBC
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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test_count_++;
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OpenOutFile(test_count_);
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char codec_ilbc[] = "ILBC";
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@ -188,9 +188,6 @@ void TestAllCodecs::Perform() {
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outfile_b_.Close();
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#endif
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#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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test_count_++;
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OpenOutFile(test_count_);
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char codec_isac[] = "ISAC";
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@ -205,9 +202,6 @@ void TestAllCodecs::Perform() {
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outfile_b_.Close();
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#endif
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#ifdef WEBRTC_CODEC_ISAC
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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test_count_++;
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OpenOutFile(test_count_);
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RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize);
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@ -220,9 +214,6 @@ void TestAllCodecs::Perform() {
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Run(channel_a_to_b_);
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outfile_b_.Close();
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#endif
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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test_count_++;
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OpenOutFile(test_count_);
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char codec_l16[] = "L16";
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@ -235,9 +226,7 @@ void TestAllCodecs::Perform() {
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RegisterSendCodec('A', codec_l16, 8000, 128000, 320, 0);
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Run(channel_a_to_b_);
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outfile_b_.Close();
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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test_count_++;
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OpenOutFile(test_count_);
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RegisterSendCodec('A', codec_l16, 16000, 256000, 160, 0);
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@ -249,9 +238,7 @@ void TestAllCodecs::Perform() {
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RegisterSendCodec('A', codec_l16, 16000, 256000, 640, 0);
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Run(channel_a_to_b_);
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outfile_b_.Close();
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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test_count_++;
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OpenOutFile(test_count_);
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RegisterSendCodec('A', codec_l16, 32000, 512000, 320, 0);
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@ -259,9 +246,7 @@ void TestAllCodecs::Perform() {
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RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0);
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Run(channel_a_to_b_);
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outfile_b_.Close();
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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test_count_++;
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OpenOutFile(test_count_);
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char codec_pcma[] = "PCMA";
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@ -277,9 +262,7 @@ void TestAllCodecs::Perform() {
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Run(channel_a_to_b_);
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RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, 0);
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Run(channel_a_to_b_);
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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char codec_pcmu[] = "PCMU";
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RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, 0);
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Run(channel_a_to_b_);
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@ -295,9 +278,6 @@ void TestAllCodecs::Perform() {
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Run(channel_a_to_b_);
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outfile_b_.Close();
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#ifdef WEBRTC_CODEC_OPUS
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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}
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test_count_++;
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OpenOutFile(test_count_);
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char codec_opus[] = "OPUS";
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@ -317,24 +297,6 @@ void TestAllCodecs::Perform() {
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Run(channel_a_to_b_);
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outfile_b_.Close();
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#endif
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if (test_mode_ != 0) {
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printf("===============================================================\n");
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/* Print out all codecs that were not tested in the run */
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printf("The following codecs was not included in the test:\n");
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#ifndef WEBRTC_CODEC_ILBC
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printf(" iLBC\n");
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#endif
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#ifndef WEBRTC_CODEC_ISAC
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printf(" ISAC float\n");
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#endif
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#ifndef WEBRTC_CODEC_ISACFX
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printf(" ISAC fix\n");
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#endif
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printf("\nTo complete the test, listen to the %d number of output files.\n",
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test_count_);
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}
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}
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// Register Codec to use in the test
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@ -354,21 +316,21 @@ void TestAllCodecs::RegisterSendCodec(char side,
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int rate,
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int packet_size,
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size_t extra_byte) {
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if (test_mode_ != 0) {
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// Print out codec and settings.
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printf("codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name,
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sampling_freq_hz, rate, packet_size);
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}
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// Store packet-size in samples, used to validate the received packet.
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// If G.722, store half the size to compensate for the timestamp bug in the
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// RFC for G.722.
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// If iSAC runs in adaptive mode, packet size in samples can change on the
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// fly, so we exclude this test by setting |packet_size_samples_| to -1.
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if (!strcmp(codec_name, "G722")) {
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int clockrate_hz = sampling_freq_hz;
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size_t num_channels = 1;
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if (absl::EqualsIgnoreCase(codec_name, "G722")) {
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packet_size_samples_ = packet_size / 2;
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} else if (!strcmp(codec_name, "ISAC") && (rate == -1)) {
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clockrate_hz = sampling_freq_hz / 2;
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} else if (absl::EqualsIgnoreCase(codec_name, "ISAC") && (rate == -1)) {
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packet_size_samples_ = -1;
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} else if (absl::EqualsIgnoreCase(codec_name, "OPUS")) {
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packet_size_samples_ = packet_size;
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num_channels = 2;
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} else {
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packet_size_samples_ = packet_size;
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}
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@ -402,16 +364,9 @@ void TestAllCodecs::RegisterSendCodec(char side,
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}
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ASSERT_TRUE(my_acm != NULL);
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// Get all codec parameters before registering
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CodecInst my_codec_param;
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CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param,
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sampling_freq_hz, 1));
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my_codec_param.rate = rate;
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my_codec_param.pacsize = packet_size;
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auto factory = CreateBuiltinAudioEncoderFactory();
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constexpr int payload_type = 17;
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SdpAudioFormat format = CodecInstToSdp(my_codec_param);
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SdpAudioFormat format = { codec_name, clockrate_hz, num_channels };
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format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
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packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
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my_acm->SetEncoder(
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@ -485,11 +440,4 @@ void TestAllCodecs::OpenOutFile(int test_number) {
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outfile_b_.Open(filename, 32000, "wb");
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}
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void TestAllCodecs::DisplaySendReceiveCodec() {
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CodecInst my_codec_param;
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printf("%s -> ", acm_a_->SendCodec()->plname);
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acm_b_->ReceiveCodec(&my_codec_param);
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printf("%s\n", my_codec_param.plname);
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}
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} // namespace webrtc
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