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.
  */
 
 #include "webrtc/api/audio_codecs/audio_format.h"
 
 #include "webrtc/common_types.h"
 
 namespace webrtc {
 
 SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
 SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
 
 SdpAudioFormat::SdpAudioFormat(const char* name,
                                int clockrate_hz,
-                               int num_channels)
+                               size_t num_channels)
     : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
 
 SdpAudioFormat::SdpAudioFormat(const std::string& name,
                                int clockrate_hz,
-                               int num_channels)
+                               size_t num_channels)
     : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
 
 SdpAudioFormat::SdpAudioFormat(const char* name,
                                int clockrate_hz,
-                               int num_channels,
+                               size_t num_channels,
                                const Parameters& param)
     : name(name),
       clockrate_hz(clockrate_hz),
       num_channels(num_channels),
       parameters(param) {}
 
 SdpAudioFormat::SdpAudioFormat(const std::string& name,
                                int clockrate_hz,
-                               int num_channels,
+                               size_t num_channels,
                                const Parameters& param)
     : name(name),
       clockrate_hz(clockrate_hz),
       num_channels(num_channels),
       parameters(param) {}
 
 SdpAudioFormat::~SdpAudioFormat() = default;
 SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
 SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
 
@@ skipping 18 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) {}
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
+                               size_t num_channels,
+                               int bitrate_bps)
+    : AudioCodecInfo(sample_rate_hz,
+                     num_channels,
+                     bitrate_bps,
+                     bitrate_bps,
+                     bitrate_bps) {}
 
-AudioCodecSpec::AudioCodecSpec(SdpAudioFormat&& format)
-    : format(std::move(format)) {}
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
+                               size_t 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_GT(sample_rate_hz, 0);
+  RTC_DCHECK_GT(num_channels, 0);
+  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);
+}
 
 }  // namespace webrtc