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}diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index 0099b2a..aad80e8 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -14,13 +14,11 @@
#include <limits>
#include <string>
+#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "common_types.h" // NOLINT(build/include)
-#include "modules/audio_coding/codecs/audio_format_conversion.h"
-#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "modules/audio_coding/test/utility.h"
+#include "modules/include/module_common_types.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/strings/string_builder.h"
@@ -35,6 +33,11 @@
// The test loops through all available mono codecs, encode at "a" sends over
// the channel, and decodes at "b".
+#define CHECK_ERROR(f) \
+ do { \
+ EXPECT_GE(f, 0) << "Error Calling API"; \
+ } while (0)
+
namespace {
const size_t kVariableSize = std::numeric_limits<size_t>::max();
}
@@ -101,7 +104,7 @@
payload_size_ = 0;
}
-TestAllCodecs::TestAllCodecs(int test_mode)
+TestAllCodecs::TestAllCodecs()
: acm_a_(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
acm_b_(AudioCodingModule::Create(
@@ -110,8 +113,6 @@
test_count_(0),
packet_size_samples_(0),
packet_size_bytes_(0) {
- // test_mode = 0 for silent test (auto test)
- test_mode_ = test_mode;
}
TestAllCodecs::~TestAllCodecs() {
@@ -126,23 +127,28 @@
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
infile_a_.Open(file_name, 32000, "rb");
- if (test_mode_ == 0) {
- RTC_LOG(LS_INFO) << "---------- TestAllCodecs ----------";
- }
-
acm_a_->InitializeReceiver();
acm_b_->InitializeReceiver();
- uint8_t num_encoders = acm_a_->NumberOfCodecs();
- CodecInst my_codec_param;
- for (uint8_t n = 0; n < num_encoders; n++) {
- acm_b_->Codec(n, &my_codec_param);
- if (!strcmp(my_codec_param.plname, "opus")) {
- my_codec_param.channels = 1;
- }
- acm_b_->RegisterReceiveCodec(my_codec_param.pltype,
- CodecInstToSdp(my_codec_param));
- }
+ acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
+ {104, {"ISAC", 32000, 1}},
+ {107, {"L16", 8000, 1}},
+ {108, {"L16", 16000, 1}},
+ {109, {"L16", 32000, 1}},
+ {111, {"L16", 8000, 2}},
+ {112, {"L16", 16000, 2}},
+ {113, {"L16", 32000, 2}},
+ {0, {"PCMU", 8000, 1}},
+ {110, {"PCMU", 8000, 2}},
+ {8, {"PCMA", 8000, 1}},
+ {118, {"PCMA", 8000, 2}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {119, {"G722", 8000, 2}},
+ {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
// Create and connect the channel
channel_a_to_b_ = new TestPack;
@@ -151,9 +157,6 @@
// All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes.
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
test_count_++;
OpenOutFile(test_count_);
char codec_g722[] = "G722";
@@ -171,9 +174,6 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#ifdef WEBRTC_CODEC_ILBC
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
test_count_++;
OpenOutFile(test_count_);
char codec_ilbc[] = "ILBC";
@@ -188,9 +188,6 @@
outfile_b_.Close();
#endif
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
test_count_++;
OpenOutFile(test_count_);
char codec_isac[] = "ISAC";
@@ -205,9 +202,6 @@
outfile_b_.Close();
#endif
#ifdef WEBRTC_CODEC_ISAC
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
test_count_++;
OpenOutFile(test_count_);
RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize);
@@ -220,9 +214,6 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
test_count_++;
OpenOutFile(test_count_);
char codec_l16[] = "L16";
@@ -235,9 +226,7 @@
RegisterSendCodec('A', codec_l16, 8000, 128000, 320, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
+
test_count_++;
OpenOutFile(test_count_);
RegisterSendCodec('A', codec_l16, 16000, 256000, 160, 0);
@@ -249,9 +238,7 @@
RegisterSendCodec('A', codec_l16, 16000, 256000, 640, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
+
test_count_++;
OpenOutFile(test_count_);
RegisterSendCodec('A', codec_l16, 32000, 512000, 320, 0);
@@ -259,9 +246,7 @@
RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
+
test_count_++;
OpenOutFile(test_count_);
char codec_pcma[] = "PCMA";
@@ -277,9 +262,7 @@
Run(channel_a_to_b_);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, 0);
Run(channel_a_to_b_);
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
+
char codec_pcmu[] = "PCMU";
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, 0);
Run(channel_a_to_b_);
@@ -295,9 +278,6 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#ifdef WEBRTC_CODEC_OPUS
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
test_count_++;
OpenOutFile(test_count_);
char codec_opus[] = "OPUS";
@@ -317,24 +297,6 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
- if (test_mode_ != 0) {
- printf("===============================================================\n");
-
- /* Print out all codecs that were not tested in the run */
- printf("The following codecs was not included in the test:\n");
-#ifndef WEBRTC_CODEC_ILBC
- printf(" iLBC\n");
-#endif
-#ifndef WEBRTC_CODEC_ISAC
- printf(" ISAC float\n");
-#endif
-#ifndef WEBRTC_CODEC_ISACFX
- printf(" ISAC fix\n");
-#endif
-
- printf("\nTo complete the test, listen to the %d number of output files.\n",
- test_count_);
- }
}
// Register Codec to use in the test
@@ -354,21 +316,21 @@
int rate,
int packet_size,
size_t extra_byte) {
- if (test_mode_ != 0) {
- // Print out codec and settings.
- printf("codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name,
- sampling_freq_hz, rate, packet_size);
- }
-
// Store packet-size in samples, used to validate the received packet.
// If G.722, store half the size to compensate for the timestamp bug in the
// RFC for G.722.
// If iSAC runs in adaptive mode, packet size in samples can change on the
// fly, so we exclude this test by setting |packet_size_samples_| to -1.
- if (!strcmp(codec_name, "G722")) {
+ int clockrate_hz = sampling_freq_hz;
+ size_t num_channels = 1;
+ if (absl::EqualsIgnoreCase(codec_name, "G722")) {
packet_size_samples_ = packet_size / 2;
- } else if (!strcmp(codec_name, "ISAC") && (rate == -1)) {
+ clockrate_hz = sampling_freq_hz / 2;
+ } else if (absl::EqualsIgnoreCase(codec_name, "ISAC") && (rate == -1)) {
packet_size_samples_ = -1;
+ } else if (absl::EqualsIgnoreCase(codec_name, "OPUS")) {
+ packet_size_samples_ = packet_size;
+ num_channels = 2;
} else {
packet_size_samples_ = packet_size;
}
@@ -402,16 +364,9 @@
}
ASSERT_TRUE(my_acm != NULL);
- // Get all codec parameters before registering
- CodecInst my_codec_param;
- CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param,
- sampling_freq_hz, 1));
- my_codec_param.rate = rate;
- my_codec_param.pacsize = packet_size;
-
auto factory = CreateBuiltinAudioEncoderFactory();
constexpr int payload_type = 17;
- SdpAudioFormat format = CodecInstToSdp(my_codec_param);
+ SdpAudioFormat format = { codec_name, clockrate_hz, num_channels };
format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
my_acm->SetEncoder(
@@ -485,11 +440,4 @@
outfile_b_.Open(filename, 32000, "wb");
}
-void TestAllCodecs::DisplaySendReceiveCodec() {
- CodecInst my_codec_param;
- printf("%s -> ", acm_a_->SendCodec()->plname);
- acm_b_->ReceiveCodec(&my_codec_param);
- printf("%s\n", my_codec_param.plname);
-}
-
} // namespace webrtc