Injectable audio encoders: BuiltinAudioEncoderFactory

webrtc/api/audio_codecs/audio_format.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 59 matching lines @@
   os << ", parameters: {";
   const char* sep = "";
   for (const auto& kv : saf.parameters) {
     os << sep << kv.first << ": " << kv.second;
     sep = ", ";
   }
   os << "}}";
   return os;
 }
 
-AudioCodecSpec::AudioCodecSpec(const SdpAudioFormat& format) : format(format) {}
+AudioFormatInfo::AudioFormatInfo()
+    : AudioFormatInfo(0, 0, 0) {}

[kwiberg-webrtc, 2017/03/15 13:33:17]

Sorry if I've already asked about this, but why is a default constructor a good
idea? It would be nicer if all AudioFormatInfo objects were in a good state.

[ossu, 2017/03/16 18:03:56]

It was necessary for one of the previous incarnations of this, but now it's not.
So I've removed it.

 
-AudioCodecSpec::AudioCodecSpec(SdpAudioFormat&& format)
-    : format(std::move(format)) {}
+AudioFormatInfo::AudioFormatInfo(int sample_rate_hz,
+                                 int num_channels,
+                                 int bitrate_bps)
+    : AudioFormatInfo(sample_rate_hz,
+                      num_channels,
+                      bitrate_bps,
+                      bitrate_bps,
+                      bitrate_bps) {}
+
+AudioFormatInfo::AudioFormatInfo(int sample_rate_hz,
+                                 int num_channels,
+                                 int default_bitrate_bps,
+                                 int min_bitrate_bps,
+                                 int max_bitrate_bps)
+    : sample_rate_hz(sample_rate_hz),
+      num_channels(num_channels),
+      default_bitrate_bps(default_bitrate_bps),
+      min_bitrate_bps(min_bitrate_bps),
+      max_bitrate_bps(max_bitrate_bps) {
+  RTC_DCHECK_GE(min_bitrate_bps, 0);
+  RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
+  RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);

[kwiberg-webrtc, 2017/03/15 13:33:17]

Also check that sample_rate_hz and num_channels are good?

[ossu, 2017/03/16 18:03:56]

Sure. I've just done the most basic of checks, that they are greater than zero.
I don't think this is the right place to enforce that an encoder is compatible
with specific sample rates or number of channels supported in WebRTC, for
example.

[kwiberg-webrtc, 2017/03/17 10:20:01]

Excellent, that was exactly what I meant.

+}
 
 }  // namespace webrtc