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Audio encoder tests: Create audio encoders the new way

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Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.

The new way of creating encoders used a 32 kbit/s bitrate
unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz
and 56 kbit/s for 32 kHz, which is what the old way of creating
encoders has used since forever.

I also had to change some test expectations on Opus, because the new
way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I
believe to be correct), while the old way defaults to 64 kbit/s in
both cases.

Bug: webrtc:8396
Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2
Reviewed-on: https://webrtc-review.googlesource.com/94540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24375}
This commit is contained in:
Karl Wiberg
2018-08-16 15:01:12 +02:00
committed by Commit Bot
parent 1165949341
commit 801500cf99
6 changed files with 76 additions and 45 deletions
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19
modules/audio_coding/acm2/acm_send_test.cc
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@ -16,10 +16,13 @@
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "rtc_base/checks.h"
#include "rtc_base/stringencode.h"
#include "test/gtest.h"
namespace webrtc {
@ -65,20 +68,26 @@ bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
sampling_freq_hz, channels));
codec.pltype = payload_type;
codec.pacsize = frame_size_samples;
codec_registered_ = (acm_->RegisterSendCodec(codec) == 0);
auto factory = CreateBuiltinAudioEncoderFactory();
SdpAudioFormat format = CodecInstToSdp(codec);
format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
frame_size_samples, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
acm_->SetEncoder(
factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
codec_registered_ = true;
input_frame_.num_channels_ = channels;
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
return codec_registered_;
}
bool AcmSendTestOldApi::RegisterExternalCodec(
AudioEncoder* external_speech_encoder) {
acm_->RegisterExternalSendCodec(external_speech_encoder);
void AcmSendTestOldApi::RegisterExternalCodec(
std::unique_ptr<AudioEncoder> external_speech_encoder) {
input_frame_.num_channels_ = external_speech_encoder->NumChannels();
acm_->SetEncoder(std::move(external_speech_encoder));
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
return codec_registered_ = true;
codec_registered_ = true;
}
std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {