ffmpeg: Add support for high resolution audio

* Output interleaved 24-bit (32-bit packed) PCM at up to 192KHz for use with PCM offload. Falls back to resampling if disabled. Change-Id: Ie1bd3066c0459fb99a9d4ff9e28d6a65e234408a
CyanogenMod · Aug 7, 2014 · 0d37b03 · 0d37b03
1 parent 45e0d61
commit 0d37b03
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Showing 3 changed files with 21 additions and 8 deletions.
diff --git a/libstagefright/FFmpegExtractor/FFmpegExtractor.cpp b/libstagefright/FFmpegExtractor/FFmpegExtractor.cpp
@@ -495,7 +495,7 @@ sp<MetaData> FFmpegExtractor::setAudioFormat(AVStream *stream)
 
         meta->setInt32(kKeyChannelCount, avctx->channels);
         meta->setInt32(kKeyBitRate, avctx->bit_rate);
-        meta->setInt32(kKeyBitspersample, avctx->bits_per_coded_sample);
+        meta->setInt32(kKeySampleBits, avctx->bits_per_coded_sample);
         meta->setInt32(kKeySampleRate, avctx->sample_rate);
         meta->setInt32(kKeyBlockAlign, avctx->block_align);
         meta->setInt32(kKeySampleFormat, avctx->sample_fmt);
@@ -1601,7 +1601,8 @@ static void adjustMPEG2TSConfidence(AVFormatContext *ic, float *confidence)
 	codec_id = getCodecId(ic, AVMEDIA_TYPE_AUDIO);
 	if (codec_id != AV_CODEC_ID_NONE
 			&& codec_id != AV_CODEC_ID_AAC
-			&& codec_id != AV_CODEC_ID_PCM_S16LE //FIXME, AV_CODEC_ID_PCM_S24LE, AV_CODEC_ID_PCM_S32LE?
+			&& codec_id != AV_CODEC_ID_PCM_S16LE
+			&& codec_id != AV_CODEC_ID_PCM_S24LE
 			&& codec_id != AV_CODEC_ID_MP1
 			&& codec_id != AV_CODEC_ID_MP2
 			&& codec_id != AV_CODEC_ID_MP3) {

diff --git a/libstagefright/codecs/ffmpegdec/adec/SoftFFmpegAudio.cpp b/libstagefright/codecs/ffmpegdec/adec/SoftFFmpegAudio.cpp
@@ -17,6 +17,7 @@
 //#define LOG_NDEBUG 0
 #define LOG_TAG "SoftFFmpegAudio"
 #include <utils/Log.h>
+#include <cutils/properties.h>
 
 #include "SoftFFmpegAudio.h"
 
@@ -92,7 +93,12 @@ SoftFFmpegAudio::SoftFFmpegAudio(
 
     setMode(name);
 
-    ALOGD("SoftFFmpegAudio component: %s mMode: %d", name, mMode);
+    char value[PROPERTY_VALUE_MAX] = {0};
+    property_get("audio.offload.24bit.enable", value, "0");
+    mHighResAudioEnabled = atoi(value);
+
+    ALOGD("SoftFFmpegAudio component: %s mMode: %d mHighResAudioEnabled: %d",
+            name, mMode, mHighResAudioEnabled);
 
     initPorts();
     CHECK_EQ(initDecoder(), (status_t)OK);
@@ -700,16 +706,20 @@ void SoftFFmpegAudio::adjustAudioParams() {
 
     channels = mCtx->channels;
 
-    //4000 <= sampling rate <= 48000
+    int32_t max_rate = mHighResAudioEnabled ? 192000 : 48000;
+
+    //4000 <= sampling rate <= 48000/192000
     if (sampling_rate < 4000) {
         sampling_rate = 4000;
-    } else if (sampling_rate > 48000) {
-        sampling_rate = 48000;
+    } else if (sampling_rate > max_rate) {
+        sampling_rate = max_rate;
     }
 
     mAudioSrcChannels = mAudioTgtChannels = channels;
     mAudioSrcFreq = mAudioTgtFreq = sampling_rate;
-    mAudioSrcFmt = mAudioTgtFmt = AV_SAMPLE_FMT_S16;
+    mAudioSrcFmt = mAudioTgtFmt =
+        (mHighResAudioEnabled && (mCtx->bits_per_coded_sample == 24))
+            ? AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S16;
     mAudioSrcChannelLayout = mAudioTgtChannelLayout =
         av_get_default_channel_layout(channels);
 }

diff --git a/libstagefright/codecs/ffmpegdec/adec/SoftFFmpegAudio.h b/libstagefright/codecs/ffmpegdec/adec/SoftFFmpegAudio.h
@@ -56,7 +56,7 @@ struct SoftFFmpegAudio : public SimpleSoftOMXComponent {
         kOutputPortIndex  = 1,
         kNumInputBuffers  = 4,
         kNumOutputBuffers = 4,
-        kOutputBufferSize = 4608 * 2
+        kOutputBufferSize = 9216 * 2
     };
 
     enum {
@@ -129,6 +129,8 @@ struct SoftFFmpegAudio : public SimpleSoftOMXComponent {
     enum AVSampleFormat mAudioSrcFmt;
     enum AVSampleFormat mAudioTgtFmt;
 
+    bool mHighResAudioEnabled;
+
     enum {
         NONE,
         AWAITING_DISABLED,