commit 7a441a0c7ddbc0ce64950157bf893f27deac5217
author kwiberg <kwiberg@webrtc.org> Tue May 31 09:46:20 2016
committer Commit bot <commit-bot@chromium.org> Tue May 31 09:46:28 2016
tree 1c24eb5f07260067b3454b73ad477ae090c61696
parent 218e16d372e05c5d290abe2daa4e23c981f5098d

AudioDecoder: New method SampleRateHz, + implementations for our codecs

This will let NetEq (and the factory, and every layer in between) keep
track of just the decoder, instead of decoder and sample rate.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2024633002
Cr-Original-Commit-Position: refs/heads/master@{#12968}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 6c2eab34f8ff1f7a724622e3473858a4c0724bd8

modules/audio_coding/codecs/g711/audio_decoder_pcm.cc

diff --git a/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
index 9757b4a..af164c4 100644
--- a/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
+++ b/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -16,6 +16,10 @@
 
 void AudioDecoderPcmU::Reset() {}
 
+int AudioDecoderPcmU::SampleRateHz() const {
+  return 8000;
+}
+
 size_t AudioDecoderPcmU::Channels() const {
   return num_channels_;
 }
@@ -25,7 +29,7 @@
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
@@ -40,6 +44,10 @@
 
 void AudioDecoderPcmA::Reset() {}
 
+int AudioDecoderPcmA::SampleRateHz() const {
+  return 8000;
+}
+
 size_t AudioDecoderPcmA::Channels() const {
   return num_channels_;
 }
@@ -49,7 +57,7 @@
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);