AudioCodingModule: Specify decoders using SdpAudioFormat

webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc

 /*
  *  Copyright (c) 2014 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/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
 
 #include <assert.h>
 #include <stdio.h>
 
 #include <memory>
 
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
@@ skipping 62 matching lines @@
   } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
     *pltype = 95;
   } else if (STR_CASE_CMP(plname, "G722") == 0) {
     *pltype = 9;
   } else {
     // Don't use any other codecs.
     return false;
   }
   return true;
 }
+
+AudioCodingModule::Config MakeAcmConfig(
+    Clock* clock,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
+  AudioCodingModule::Config config;
+  config.id = 0;
+  config.clock = clock;
+  config.decoder_factory = std::move(decoder_factory);
+  return config;
+}
+
 }  // namespace
 
 AcmReceiveTestOldApi::AcmReceiveTestOldApi(
     PacketSource* packet_source,
     AudioSink* audio_sink,
     int output_freq_hz,
-    NumOutputChannels exptected_output_channels)
+    NumOutputChannels exptected_output_channels,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
     : clock_(0),
-      acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
+      acm_(webrtc::AudioCodingModule::Create(
+          MakeAcmConfig(&clock_, std::move(decoder_factory)))),
       packet_source_(packet_source),
       audio_sink_(audio_sink),
       output_freq_hz_(output_freq_hz),
-      exptected_output_channels_(exptected_output_channels) {
-}
+      exptected_output_channels_(exptected_output_channels) {}
 
 AcmReceiveTestOldApi::~AcmReceiveTestOldApi() = default;
 
 void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
   CodecInst my_codec_param;
   for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
     ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
     if (ModifyAndUseThisCodec(&my_codec_param)) {
       ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
           << "Couldn't register receive codec.\n";
@@ skipping 78 matching lines @@
 AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
     PacketSource* packet_source,
     AudioSink* audio_sink,
     int output_freq_hz_1,
     int output_freq_hz_2,
     int toggle_period_ms,
     NumOutputChannels exptected_output_channels)
     : AcmReceiveTestOldApi(packet_source,
                            audio_sink,
                            output_freq_hz_1,
-                           exptected_output_channels),
+                           exptected_output_channels,
+                           CreateBuiltinAudioDecoderFactory()),
       output_freq_hz_1_(output_freq_hz_1),
       output_freq_hz_2_(output_freq_hz_2),
       toggle_period_ms_(toggle_period_ms),
-      last_toggle_time_ms_(clock_.TimeInMilliseconds()) {
-}
+      last_toggle_time_ms_(clock_.TimeInMilliseconds()) {}
 
 void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
   if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
     output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
                           ? output_freq_hz_2_
                           : output_freq_hz_1_;
     last_toggle_time_ms_ = clock_.TimeInMilliseconds();
   }
 }
 
 }  // namespace test
 }  // namespace webrtc