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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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4
api/audio_codecs/isac/audio_encoder_isac_float.cc
View File

@ -24,6 +24,7 @@ AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
format.num_channels == 1) {
Config config;
config.sample_rate_hz = format.clockrate_hz;
config.bit_rate = format.clockrate_hz == 16000 ? 32000 : 56000;
if (config.sample_rate_hz == 16000) {
// For sample rate 16 kHz, optionally use 60 ms frames, instead of the
// default 30 ms.
@ -65,9 +66,10 @@ std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder(
absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
AudioEncoderIsacFloatImpl::Config c;
c.payload_type = payload_type;
c.sample_rate_hz = config.sample_rate_hz;
c.frame_size_ms = config.frame_size_ms;
c.payload_type = payload_type;
c.bit_rate = config.bit_rate;
return absl::make_unique<AudioEncoderIsacFloatImpl>(c);
}