Injectable audio encoders: BuiltinAudioEncoderFactory

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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/codecs/g711/audio_encoder_pcm.h"
 
 #include <limits>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/string_to_number.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 
 namespace webrtc {
 
 namespace {
 
 template <typename T>
 typename T::Config CreateConfig(const CodecInst& codec_inst) {
   typename T::Config config;
   config.frame_size_ms = codec_inst.pacsize / 8;
   config.num_channels = codec_inst.channels;
   config.payload_type = codec_inst.pltype;
   return config;
 }
 
+template <typename T>
+typename T::Config CreateConfig(int payload_type,
+                                const SdpAudioFormat& format) {
+  typename T::Config config;
+  config.frame_size_ms = 20;
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      config.frame_size_ms = *ptime;
+    }
+  }
+  config.num_channels = format.num_channels;
+  config.payload_type = payload_type;
+  return config;
+}
+
+template <typename T>
+rtc::Optional<AudioFormatInfo> QueryAudioFormatImpl(
+    const SdpAudioFormat& format) {
+  if (STR_CASE_CMP(format.name.c_str(), T::GetPayloadName()) == 0 &&
+      format.clockrate_hz == 8000 && format.num_channels >= 1) {
+    typename T::Config config = CreateConfig<T>(0, format);
+    if (config.IsOk()) {
+      return rtc::Optional<AudioFormatInfo>({8000, format.num_channels, 64000});
+    }
+  }
+  return rtc::Optional<AudioFormatInfo>();
+}
+
+
 }  // namespace
 
 bool AudioEncoderPcm::Config::IsOk() const {
   return (frame_size_ms % 10 == 0) && (num_channels >= 1);
 }
 
 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
     : sample_rate_hz_(sample_rate_hz),
       num_channels_(config.num_channels),
       payload_type_(config.payload_type),
@@ skipping 58 matching lines @@
   return info;
 }
 
 void AudioEncoderPcm::Reset() {
   speech_buffer_.clear();
 }
 
 AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst)
     : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {}
 
+AudioEncoderPcmA::AudioEncoderPcmA(int payload_type,
+                                   const SdpAudioFormat& format)
+    : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(payload_type, format)) {}
+
+rtc::Optional<AudioFormatInfo> AudioEncoderPcmA::QueryAudioFormat(
+    const SdpAudioFormat& format) {
+  return QueryAudioFormatImpl<AudioEncoderPcmA>(format);
+}
+
 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeA(audio, input_len, encoded);
 }
 
 size_t AudioEncoderPcmA::BytesPerSample() const {
   return 1;
 }
 
 AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
   return AudioEncoder::CodecType::kPcmA;
 }
 
 AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst)
     : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {}
 
+AudioEncoderPcmU::AudioEncoderPcmU(int payload_type,
+                                   const SdpAudioFormat& format)
+    : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(payload_type, format)) {}
+
+rtc::Optional<AudioFormatInfo> AudioEncoderPcmU::QueryAudioFormat(
+    const SdpAudioFormat& format) {
+  return QueryAudioFormatImpl<AudioEncoderPcmU>(format);
+}
+
 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
 size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
 AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
   return AudioEncoder::CodecType::kPcmU;
 }
 
 }  // namespace webrtc