Injectable audio encoders: BuiltinAudioEncoderFactory

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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/g722/audio_encoder_g722.h"
 
+#include <algorithm>
+
 #include <limits>
 #include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/string_to_number.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
 namespace {
 
 const size_t kSampleRateHz = 16000;
 
 AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderG722::Config config;
   config.num_channels = codec_inst.channels;
   config.frame_size_ms = codec_inst.pacsize / 16;
   config.payload_type = codec_inst.pltype;
   return config;
 }
 
+AudioEncoderG722::Config CreateConfig(int payload_type,
+                                      const SdpAudioFormat& format) {
+  AudioEncoderG722::Config config;
+  config.payload_type = payload_type;
+  config.num_channels = format.num_channels;
+  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) {
+      const int whole_packets = *ptime / 10;
+      config.frame_size_ms = std::max(10, std::min(whole_packets * 10, 60));
+    }
+  }
+  return config;
+}
+
 }  // namespace
 
 bool AudioEncoderG722::Config::IsOk() const {
   return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) &&
       (num_channels >= 1);
 }
 
 AudioEncoderG722::AudioEncoderG722(const Config& config)
     : num_channels_(config.num_channels),
       payload_type_(config.payload_type),
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
       num_10ms_frames_buffered_(0),
       first_timestamp_in_buffer_(0),
       encoders_(new EncoderState[num_channels_]),
       interleave_buffer_(2 * num_channels_) {
   RTC_CHECK(config.IsOk());
   const size_t samples_per_channel =
       kSampleRateHz / 100 * num_10ms_frames_per_packet_;
   for (size_t i = 0; i < num_channels_; ++i) {
     encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
     encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
   }
   Reset();
 }
 
 AudioEncoderG722::AudioEncoderG722(const CodecInst& codec_inst)
     : AudioEncoderG722(CreateConfig(codec_inst)) {}
 
+AudioEncoderG722::AudioEncoderG722(int payload_type,
+                                   const SdpAudioFormat& format)
+    : AudioEncoderG722(CreateConfig(payload_type, format)) {}
+
 AudioEncoderG722::~AudioEncoderG722() = default;
 
+rtc::Optional<AudioCodecInfo> AudioEncoderG722::QueryAudioEncoder(
+    const SdpAudioFormat& format) {
+  if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) {
+    Config config = CreateConfig(0, format);
+    if (format.clockrate_hz == 8000 && config.IsOk()) {
+      return rtc::Optional<AudioCodecInfo>(
+          {rtc::dchecked_cast<int>(kSampleRateHz), config.num_channels, 64000});
+    }
+  }
+  return rtc::Optional<AudioCodecInfo>();
+}
+
 int AudioEncoderG722::SampleRateHz() const {
   return kSampleRateHz;
 }
 
 size_t AudioEncoderG722::NumChannels() const {
   return num_channels_;
 }
 
 int AudioEncoderG722::RtpTimestampRateHz() const {
   // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
@@ skipping 82 matching lines @@
 
 AudioEncoderG722::EncoderState::~EncoderState() {
   RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
 }
 
 size_t AudioEncoderG722::SamplesPerChannel() const {
   return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
 }
 
 }  // namespace webrtc