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}

api/audio_codecs/isac/audio_encoder_isac_float.cc

@@ -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);
 }
 

api/audio_codecs/isac/audio_encoder_isac_float.h

@@ -28,12 +28,30 @@ namespace webrtc {
 struct AudioEncoderIsacFloat {
   struct Config {
     bool IsOk() const {
-      return (sample_rate_hz == 16000 &&
-              (frame_size_ms == 30 || frame_size_ms == 60)) ||
-             (sample_rate_hz == 32000 && frame_size_ms == 30);
+      switch (sample_rate_hz) {
+        case 16000:
+          if (frame_size_ms != 30 && frame_size_ms != 60) {
+            return false;
+          }
+          if (bit_rate < 10000 || bit_rate > 32000) {
+            return false;
+          }
+          return true;
+        case 32000:
+          if (frame_size_ms != 30) {
+            return false;
+          }
+          if (bit_rate < 10000 || bit_rate > 56000) {
+            return false;
+          }
+          return true;
+        default:
+          return false;
+      }
     }
     int sample_rate_hz = 16000;
     int frame_size_ms = 30;
+    int bit_rate = 32000;  // Limit on short-term average bit rate, in bits/s.
   };
   static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);