From 133cff009b7677af4649c282f1e2546da4dba975 Mon Sep 17 00:00:00 2001 From: Karl Wiberg Date: Fri, 6 Jul 2018 15:40:14 +0200 Subject: [PATCH] AudioCodingModuleTest.TestAllCodecs: Create audio encoders the new way Specifically, don't expect the ACM to be able to create encoders; we have to give it an encoder that we make ourselves. To make it work, I had to add support for the "ptime" parameter to the L16 codec. Bug: webrtc:8396 Change-Id: I3869422882611d2eed65d6c849ea7cd3ad6bd126 Reviewed-on: https://webrtc-review.googlesource.com/87423 Reviewed-by: Oskar Sundbom Commit-Queue: Karl Wiberg Cr-Commit-Position: refs/heads/master@{#24217} --- api/audio_codecs/L16/BUILD.gn | 1 + api/audio_codecs/L16/audio_encoder_L16.cc | 9 +++++++++ modules/audio_coding/BUILD.gn | 1 + modules/audio_coding/test/TestAllCodecs.cc | 11 ++++++++++- 4 files changed, 21 insertions(+), 1 deletion(-) diff --git a/api/audio_codecs/L16/BUILD.gn b/api/audio_codecs/L16/BUILD.gn index b1160d4e34..9724805001 100644 --- a/api/audio_codecs/L16/BUILD.gn +++ b/api/audio_codecs/L16/BUILD.gn @@ -24,6 +24,7 @@ rtc_static_library("audio_encoder_L16") { "../../..:webrtc_common", "../../../modules/audio_coding:pcm16b", "../../../rtc_base:rtc_base_approved", + "../../../rtc_base:safe_minmax", "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/types:optional", ] diff --git a/api/audio_codecs/L16/audio_encoder_L16.cc b/api/audio_codecs/L16/audio_encoder_L16.cc index 35a3b9870f..b516f62863 100644 --- a/api/audio_codecs/L16/audio_encoder_L16.cc +++ b/api/audio_codecs/L16/audio_encoder_L16.cc @@ -15,6 +15,8 @@ #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h" #include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/numerics/safe_minmax.h" +#include "rtc_base/string_to_number.h" namespace webrtc { @@ -26,6 +28,13 @@ absl::optional AudioEncoderL16::SdpToConfig( Config config; config.sample_rate_hz = format.clockrate_hz; config.num_channels = rtc::dchecked_cast(format.num_channels); + auto ptime_iter = format.parameters.find("ptime"); + if (ptime_iter != format.parameters.end()) { + const auto ptime = rtc::StringToNumber(ptime_iter->second); + if (ptime && *ptime > 0) { + config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60); + } + } return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk() ? absl::optional(config) : absl::nullopt; diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 8cda506132..380f1bc740 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -1327,6 +1327,7 @@ if (rtc_include_tests) { "../..:webrtc_common", "../../api/audio:audio_frame_api", "../../api/audio_codecs:builtin_audio_decoder_factory", + "../../api/audio_codecs:builtin_audio_encoder_factory", "../../rtc_base:checks", "../../rtc_base:rtc_base_approved", "../../rtc_base/synchronization:rw_lock_wrapper", diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc index f78b705b7b..781337c107 100644 --- a/modules/audio_coding/test/TestAllCodecs.cc +++ b/modules/audio_coding/test/TestAllCodecs.cc @@ -15,12 +15,14 @@ #include #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 "rtc_base/logging.h" +#include "rtc_base/stringencode.h" #include "test/gtest.h" #include "test/testsupport/fileutils.h" @@ -405,7 +407,14 @@ void TestAllCodecs::RegisterSendCodec(char side, sampling_freq_hz, 1)); my_codec_param.rate = rate; my_codec_param.pacsize = packet_size; - CHECK_ERROR(my_acm->RegisterSendCodec(my_codec_param)); + + auto factory = CreateBuiltinAudioEncoderFactory(); + constexpr int payload_type = 17; + SdpAudioFormat format = CodecInstToSdp(my_codec_param); + format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact( + packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000))); + my_acm->SetEncoder( + factory->MakeAudioEncoder(payload_type, format, absl::nullopt)); } void TestAllCodecs::Run(TestPack* channel) {