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.
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}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 2829db4..de2aeb7 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1406,8 +1406,10 @@
"../../api/audio:audio_frame_api",
"../../rtc_base:checks",
":audio_coding",
+ ":audio_format_conversion",
":neteq_tools",
"../../api/audio_codecs:builtin_audio_decoder_factory",
+ "../../api/audio_codecs:builtin_audio_encoder_factory",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:rtc_base_approved",
"../../test:test_support",