Injectable audio encoders: BuiltinAudioEncoderFactory

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

Index: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index e4d58ae4d40e536fe25051167ea4401655206185..9a980e4cea2ce9b9f36839bf1b94cb3f2aa2e568 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -12,9 +12,15 @@
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/string_to_number.h"
 #include "webrtc/common_types.h"
 
 namespace webrtc {
+namespace {
+int GetIsacMaxBitrate(int clockrate_hz) {
+  return (clockrate_hz == 32000) ? 56000 : 32000;
+}
+}  // namespace
 
 template <typename T>
 typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
@@ -33,6 +39,33 @@ typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
 }
 
 template <typename T>
+typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
+    int payload_type,
+    const SdpAudioFormat& format) {
+  typename AudioEncoderIsacT<T>::Config config;
+  config.payload_type = payload_type;
+  config.sample_rate_hz = format.clockrate_hz;
+
+  // We only support different frame sizes at 16000 kHz.

[the sun, 2017/03/20 20:53:40]

That rate is *way* too high. :P

[ossu, 2017/03/21 16:15:31]

:D

+  if (config.sample_rate_hz == 16000) {
+    auto ptime_iter = format.parameters.find("ptime");
+    if (ptime_iter != format.parameters.end()) {
+      auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+      if (ptime && *ptime >= 60) {

[ossu, 2017/03/20 18:18:53]

The ptime parsers are based on the values found in kCodecPrefs in
webrtcvoiceengine.cc.

[the sun, 2017/03/20 20:53:40]

Acknowledged.

+        config.frame_size_ms = 60;
+      } else {
+        config.frame_size_ms = 30;
+      }
+    }
+  }
+
+  // Set the default bitrate for ISAC to the maximum bitrate allowed at this
+  // clockrate. At this point, adaptive mode is not used by WebRTC.
+  config.bit_rate = GetIsacMaxBitrate(format.clockrate_hz);
+  return config;
+}
+
+template <typename T>
 bool AudioEncoderIsacT<T>::Config::IsOk() const {
   if (max_bit_rate < 32000 && max_bit_rate != -1)
     return false;
@@ -73,6 +106,25 @@ AudioEncoderIsacT<T>::AudioEncoderIsacT(
     : AudioEncoderIsacT(CreateIsacConfig<T>(codec_inst, bwinfo)) {}
 
 template <typename T>
+AudioEncoderIsacT<T>::AudioEncoderIsacT(int payload_type,
+                                        const SdpAudioFormat& format)
+    : AudioEncoderIsacT(CreateIsacConfig<T>(payload_type, format)) {}
+
+template <typename T>
+rtc::Optional<AudioCodecInfo> AudioEncoderIsacT<T>::QueryAudioEncoder(
+    const SdpAudioFormat& format) {
+  if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) {
+    Config config = CreateIsacConfig<T>(0, format);
+    if (config.IsOk()) {
+      return rtc::Optional<AudioCodecInfo>({
+          config.sample_rate_hz, 1, config.bit_rate, 10000,
+          GetIsacMaxBitrate(format.clockrate_hz)});
+    }
+  }
+  return rtc::Optional<AudioCodecInfo>();
+}
+
+template <typename T>
 AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
   RTC_CHECK_EQ(0, T::Free(isac_state_));
 }