| Index: webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
|
| diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
|
| index cafd3e851bd376cf37aef2a9633b7bfa887fbcd7..85424745d8e4f394ec54757cafec05baf0b61474 100644
|
| --- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
|
| +++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
|
| @@ -11,6 +11,7 @@
|
| #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
|
|
|
| #include "webrtc/base/checks.h"
|
| +#include "webrtc/base/string_to_number.h"
|
| #include "webrtc/common_types.h"
|
| #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
|
|
|
| @@ -40,6 +41,23 @@ AudioEncoderPcm16B::Config CreateConfig(const CodecInst& codec_inst) {
|
| config.payload_type = codec_inst.pltype;
|
| return config;
|
| }
|
| +
|
| +AudioEncoderPcm16B::Config CreateConfig(int payload_type,
|
| + const SdpAudioFormat& format) {
|
| + AudioEncoderPcm16B::Config config;
|
| + config.num_channels = format.num_channels;
|
| + config.sample_rate_hz = format.clockrate_hz;
|
| + config.frame_size_ms = 10;
|
| + 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.payload_type = payload_type;
|
| + return config;
|
| +}
|
| } // namespace
|
|
|
| bool AudioEncoderPcm16B::Config::IsOk() const {
|
| @@ -52,4 +70,22 @@ bool AudioEncoderPcm16B::Config::IsOk() const {
|
| AudioEncoderPcm16B::AudioEncoderPcm16B(const CodecInst& codec_inst)
|
| : AudioEncoderPcm16B(CreateConfig(codec_inst)) {}
|
|
|
| +AudioEncoderPcm16B::AudioEncoderPcm16B(int payload_type,
|
| + const SdpAudioFormat& format)
|
| + : AudioEncoderPcm16B(CreateConfig(payload_type, format)) {}
|
| +
|
| +rtc::Optional<AudioFormatInfo> AudioEncoderPcm16B::QueryAudioFormat(
|
| + const SdpAudioFormat& format) {
|
| + if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
|
| + format.num_channels >= 1) {
|
| + Config config = CreateConfig(0, format);
|
| + if (config.IsOk()) {
|
| + return rtc::Optional<AudioFormatInfo>(
|
| + {config.sample_rate_hz, config.num_channels,
|
| + config.sample_rate_hz * 2 * config.num_channels});
|
| + }
|
| + }
|
| + return rtc::Optional<AudioFormatInfo>();
|
| +}
|
| +
|
| } // namespace webrtc
|
|
|