commit f693bfae5ff0a031b4ceeed13655527b411ce650
author Fredrik Solenberg <solenberg@webrtc.org> Tue Dec 11 12:22:10 2018 +0100
committer Commit Bot <commit-bot@chromium.org> Mon Dec 17 10:33:55 2018 +0000
tree 899f0cd708c381a101c090fe1d3c9496e6a85e77
parent a134204aa3ab2b09fd109c59bc3472c92d9e0ae6

Remove CodecInst pt.2

The following APIs on AudioCodingModule are deprecated with this CL:
  static int NumberOfCodecs();
  static int Codec(int, CodecInst*);
  static int Codec(const char*, CodecInst*, int, size_t);
  static int Codec(const char*, int, size_t);
  absl::optional<CodecInst> SendCodec() const;
  bool RegisterReceiveCodec(int, const SdpAudioFormat&);
  int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
  int UnregisterReceiveCodec(uint8_t);
  int32_t ReceiveCodec(CodecInst*);
  absl::optional<SdpAudioFormat> ReceiveFormat();

As well as this method on RtpRtcp module:
  int32_t RegisterSendPayload(const CodecInst&);

Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}

modules/audio_coding/acm2/acm_send_test.cc

diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index c5e010c..cf9c8eb 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -14,10 +14,10 @@
 #include <stdio.h>
 #include <string.h>
 
+#include "absl/strings/match.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "modules/audio_coding/codecs/audio_format_conversion.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "modules/audio_coding/neteq/tools/packet.h"
@@ -59,23 +59,28 @@
 AcmSendTestOldApi::~AcmSendTestOldApi() = default;
 
 bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
-                                      int sampling_freq_hz,
-                                      int channels,
+                                      int clockrate_hz,
+                                      int num_channels,
                                       int payload_type,
                                       int frame_size_samples) {
-  CodecInst codec;
-  RTC_CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec,
-                                           sampling_freq_hz, channels));
-  codec.pltype = payload_type;
-  codec.pacsize = frame_size_samples;
-  auto factory = CreateBuiltinAudioEncoderFactory();
-  SdpAudioFormat format = CodecInstToSdp(codec);
+  SdpAudioFormat format(payload_name, clockrate_hz, num_channels);
+  if (absl::EqualsIgnoreCase(payload_name, "g722")) {
+    RTC_CHECK_EQ(16000, clockrate_hz);
+    format.clockrate_hz = 8000;
+  } else if (absl::EqualsIgnoreCase(payload_name, "opus")) {
+    RTC_CHECK(num_channels == 1 || num_channels == 2);
+    if (num_channels == 2) {
+      format.parameters["stereo"] = "1";
+    }
+    format.num_channels = 2;
+  }
   format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
-      frame_size_samples, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
+      frame_size_samples, rtc::CheckedDivExact(clockrate_hz, 1000)));
+  auto factory = CreateBuiltinAudioEncoderFactory();
   acm_->SetEncoder(
       factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
   codec_registered_ = true;
-  input_frame_.num_channels_ = channels;
+  input_frame_.num_channels_ = num_channels;
   assert(input_block_size_samples_ * input_frame_.num_channels_ <=
          AudioFrame::kMaxDataSizeSamples);
   return codec_registered_;