Audio encoder tests: 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.

Bug: webrtc:8396
Change-Id: I032b12f3813af6ac3ea0dfb688006899dffe4855
Reviewed-on: https://webrtc-review.googlesource.com/94150
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24323}
This commit is contained in:
Karl Wiberg
2018-08-15 15:20:49 +02:00
committed by Commit Bot
parent d2e9c59a4b
commit 658a552fd5
2 changed files with 10 additions and 5 deletions

View File

@ -16,6 +16,7 @@
#include <sstream> // no-presubmit-check TODO(webrtc:8982)
#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"
@ -74,7 +75,8 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
sendCodec.channels = channels;
EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
acm->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
sendCodec.pltype, CodecInstToSdp(sendCodec), absl::nullopt));
_packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
EXPECT_EQ(0, acm->RegisterTransportCallback(_packetization));