Injectable audio encoders: BuiltinAudioEncoderFactory

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.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/ilbc/audio_encoder_ilbc.h"
 
 #include <algorithm>
 #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/ilbc/ilbc.h"
 
 namespace webrtc {
 
 namespace {
 
 const int kSampleRateHz = 8000;
 
 AudioEncoderIlbc::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderIlbc::Config config;
   config.frame_size_ms = codec_inst.pacsize / 8;
   config.payload_type = codec_inst.pltype;
   return config;
 }
 
+AudioEncoderIlbc::Config CreateConfig(int payload_type,
+                                      const SdpAudioFormat& format) {
+  AudioEncoderIlbc::Config config;
+  config.payload_type = payload_type;
+  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;
+    }
+  }
+  return config;
+}
+
+int GetIlbcBitrate(int ptime) {
+  switch (ptime) {
+    case 20: case 40:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 30: case 60:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      FATAL();
+  }
+}
+
 }  // namespace
 
 // static
 const size_t AudioEncoderIlbc::kMaxSamplesPerPacket;
 
 bool AudioEncoderIlbc::Config::IsOk() const {
   return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
           frame_size_ms == 60) &&
       static_cast<size_t>(kSampleRateHz / 100 * (frame_size_ms / 10)) <=
           kMaxSamplesPerPacket;
 }
 
 AudioEncoderIlbc::AudioEncoderIlbc(const Config& config)
     : config_(config),
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
       encoder_(nullptr) {
   Reset();
 }
 
 AudioEncoderIlbc::AudioEncoderIlbc(const CodecInst& codec_inst)
     : AudioEncoderIlbc(CreateConfig(codec_inst)) {}
 
+AudioEncoderIlbc::AudioEncoderIlbc(int payload_type,
+                                   const SdpAudioFormat& format)
+    : AudioEncoderIlbc(CreateConfig(payload_type, format)) {}
+
+rtc::Optional<AudioFormatInfo> AudioEncoderIlbc::QueryAudioFormat(
+    const SdpAudioFormat& format) {
+  if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
+      format.clockrate_hz == 8000 && format.num_channels == 1) {
+    Config config = CreateConfig(0, format);
+    if (config.IsOk()) {
+      return rtc::Optional<AudioFormatInfo>(
+          {kSampleRateHz, 1, GetIlbcBitrate(config.frame_size_ms)});
+    }
+  }
+
+  return rtc::Optional<AudioFormatInfo>();
+}
+
+
 AudioEncoderIlbc::~AudioEncoderIlbc() {
   RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
 }
 
 int AudioEncoderIlbc::SampleRateHz() const {
   return kSampleRateHz;
 }
 
 size_t AudioEncoderIlbc::NumChannels() const {
   return 1;
 }
 
 size_t AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 size_t AudioEncoderIlbc::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderIlbc::GetTargetBitrate() const {
-  switch (num_10ms_frames_per_packet_) {
-    case 2: case 4:
-      // 38 bytes per frame of 20 ms => 15200 bits/s.
-      return 15200;
-    case 3: case 6:
-      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
-      return 13333;
-    default:
-      FATAL();
-  }
+  return GetIlbcBitrate(num_10ms_frames_per_packet_ * 10);
 }
 
 AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
 
   // Save timestamp if starting a new packet.
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
@@ skipping 51 matching lines @@
   switch (num_10ms_frames_per_packet_) {
     case 2:   return 38;
     case 3:   return 50;
     case 4:   return 2 * 38;
     case 6:   return 2 * 50;
     default:  FATAL();
   }
 }
 
 }  // namespace webrtc