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..bbe40a4b196ab759560c7961b9e5f2cbeee29131 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 |
@@ -12,6 +12,7 @@ |
#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 { |
@@ -33,6 +34,28 @@ 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; |
+ |
+ 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; |
+ } |
+ } |
+ |
+ // TODO(ossu): These values are taken from ACMCodecDB. At this |
+ // point, adaptive mode is not used by WebRTC. |
+ config.bit_rate = (format.clockrate_hz == 32000) ? 56000 : 32000; |
+ return config; |
+} |
+ |
+template <typename T> |
bool AudioEncoderIsacT<T>::Config::IsOk() const { |
if (max_bit_rate < 32000 && max_bit_rate != -1) |
return false; |
@@ -73,6 +96,25 @@ 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<AudioFormatInfo> AudioEncoderIsacT<T>::QueryAudioFormat( |
+ 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<AudioFormatInfo>({ |
+ config.sample_rate_hz, 1, config.bit_rate, 10000, |
+ (config.sample_rate_hz == 16000) ? 32000 : 56000}); |
+ } |
+ } |
+ return rtc::Optional<AudioFormatInfo>(); |
+} |
+ |
+template <typename T> |
AudioEncoderIsacT<T>::~AudioEncoderIsacT() { |
RTC_CHECK_EQ(0, T::Free(isac_state_)); |
} |