hardknott: tinycompress: sync with meta-imx hardknott-5.10.72-2.2.0 branch
Add missing patch Signed-off-by: Gabriel Valcazar <gabriel.valcazar@digi.com>
This commit is contained in:
Gabriel Valcazar
parent
157db7a51d
commit
030fc2a641
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@ -0,0 +1,252 @@
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From 2912f8573cea25fbd38ac7a8b68af2ea6a05e599 Mon Sep 17 00:00:00 2001
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From: Zhang Peng <peng.zhang_8@nxp.com>
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Date: Wed, 28 Oct 2020 19:08:53 +0800
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Subject: [PATCH] cplay: Support aac streams
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Support run aac format streams for cplay.
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upstream status: imx specific
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Signed-off-by: Zhang Peng <peng.zhang_8@nxp.com>
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---
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src/utils/cplay.c | 210 ++++++++++++++++++++++++++++++++++++++++++++++
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1 file changed, 210 insertions(+)
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diff --git a/src/utils/cplay.c b/src/utils/cplay.c
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index 8e3dcbb..2a1464a 100644
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--- a/src/utils/cplay.c
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+++ b/src/utils/cplay.c
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@@ -245,6 +245,190 @@ static int parse_wav_header(FILE *file, unsigned int *num_channels, unsigned int
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return 0;
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}
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+int find_adts_header(FILE *file, unsigned int *num_channels, unsigned int *sample_rate, unsigned int *format)
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+{
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+ int ret;
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+ unsigned char buf[5];
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+
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+ ret = fread(buf, sizeof(buf), 1, file);
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+ if (ret < 0) {
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+ fprintf(stderr, "open file error: %d\n", ret);
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+ return 0;
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+ }
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+ fseek(file, 0, SEEK_SET);
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+
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+ if ((buf[0] != 0xff) || (buf[1] & 0xf0 != 0xf0))
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+ return 0;
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+ /* mpeg id */
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+ switch (buf[1]>>3 & 0x1) {
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+ case 0x0:
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+ *format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
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+ break;
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+ case 0x1:
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+ *format = SND_AUDIOSTREAMFORMAT_MP2ADTS;
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+ break;
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+ default:
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+ fprintf(stderr, "can't find stream format\n");
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+ break;
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+ }
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+ /* sample_rate */
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+ switch (buf[2]>>2 & 0xf) {
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+ case 0x0:
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+ *sample_rate = 96000;
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+ break;
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+ case 0x1:
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+ *sample_rate = 88200;
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+ break;
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+ case 0x2:
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+ *sample_rate = 64000;
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+ break;
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+ case 0x3:
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+ *sample_rate = 48000;
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+ break;
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+ case 0x4:
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+ *sample_rate = 44100;
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+ break;
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+ case 0x5:
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+ *sample_rate = 32000;
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+ break;
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+ case 0x6:
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+ *sample_rate = 24000;
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+ break;
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+ case 0x7:
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+ *sample_rate = 22050;
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+ break;
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+ case 0x8:
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+ *sample_rate = 16000;
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+ break;
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+ case 0x9:
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+ *sample_rate = 12000;
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+ break;
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+ case 0xa:
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+ *sample_rate = 11025;
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+ break;
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+ case 0xb:
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+ *sample_rate = 8000;
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+ break;
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+ case 0xc:
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+ *sample_rate = 7350;
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+ break;
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+ default:
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+ break;
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+ }
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+ /* channel */
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+ switch (((buf[2]&0x1) << 2) | (buf[3]>>6)) {
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+ case 1:
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+ *num_channels = 1;
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+ break;
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+ case 2:
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+ *num_channels = 2;
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+ break;
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+ case 3:
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+ *num_channels = 3;
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+ break;
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+ case 4:
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+ *num_channels = 4;
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+ break;
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+ case 5:
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+ *num_channels = 5;
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+ break;
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+ case 6:
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+ *num_channels = 6;
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+ break;
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+ case 7:
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+ *num_channels = 7;
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+ break;
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+ default:
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+ break;
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+ }
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+ return 1;
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+}
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+
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+static const int aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100,
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+ 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350
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+};
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+
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+#define MAX_SR_NUM sizeof(aac_sample_rates)/sizeof(aac_sample_rates[0])
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+
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+static int get_sample_rate_from_index(int sr_index)
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+{
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+ if (sr_index >= 0 && sr_index < MAX_SR_NUM)
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+ return aac_sample_rates[sr_index];
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+
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+ return 0;
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+}
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+
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+int find_adif_header(FILE *file, unsigned int *num_channels, unsigned int *sample_rate, unsigned int *format)
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+{
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+ int ret;
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+ unsigned char adif_id[4];
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+ unsigned char adif_header[20];
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+ int bitstream_type;
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+ int bitrate;
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+ int object_type;
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+ int sr_index;
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+ int skip_size = 0;
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+
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+ ret = fread(adif_id, sizeof(unsigned char), 4, file);
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+ if (ret < 0) {
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+ fprintf(stderr, "read data from file err: %d\n", ret);
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+ return 0;
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+ }
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+ /* adif id */
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+ if ((adif_id[0] != 0x41) || (adif_id[1] != 0x44) ||
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+ (adif_id[2] != 0x49) || (adif_id[3] != 0x46))
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+ return 0;
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+
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+ fread(adif_header, sizeof(unsigned char), 20, file);
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+
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+ /* copyright string */
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+ if (adif_header[0] & 0x80)
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+ skip_size = 9;
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+
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+ bitstream_type = adif_header[0 + skip_size] & 0x10;
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+ bitrate =
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+ ((unsigned int) (adif_header[0 + skip_size] & 0x0f) << 19) |
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+ ((unsigned int) adif_header[1 + skip_size] << 11) |
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+ ((unsigned int) adif_header[2 + skip_size] << 3) |
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+ ((unsigned int) adif_header[3 + skip_size] & 0xe0);
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+
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+ if (bitstream_type == 0) {
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+ object_type = ((adif_header[6 + skip_size] & 0x01) << 1) |
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+ ((adif_header[7 + skip_size] & 0x80) >> 7);
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+ sr_index = (adif_header[7 + skip_size] & 0x78) >> 3;
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+ }
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+ /* VBR */
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+ else {
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+ object_type = (adif_header[4 + skip_size] & 0x18) >> 3;
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+ sr_index = ((adif_header[4 + skip_size] & 0x07) << 1) |
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+ ((adif_header[5 + skip_size] & 0x80) >> 7);
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+ }
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+
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+ /* sample rate */
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+ *sample_rate = get_sample_rate_from_index(sr_index);
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+
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+ /* FIXME: assume channels is 2 */
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+ *num_channels = 2;
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+
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+ *format = SND_AUDIOSTREAMFORMAT_ADIF;
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+ fseek(file, 0, SEEK_SET);
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+ return 1;
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+}
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+
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+static int parse_aac_header(FILE *file, unsigned int *num_channels, unsigned int *sample_rate, unsigned int *format)
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+{
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+ if (find_adts_header(file, num_channels, sample_rate, format))
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+ return 1;
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+ else if (find_adif_header(file, num_channels, sample_rate, format))
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+ return 1;
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+ else {
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+ fprintf(stderr, "can't find streams format\n");
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+ return 0;
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+ }
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+
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+ return 1;
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+}
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+
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static int print_time(struct compress *compress)
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{
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unsigned int avail;
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@@ -513,6 +697,29 @@ void get_codec_pcm(FILE *file, struct compr_config *config,
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codec->format = format;
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}
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+void get_codec_aac(FILE *file, struct compr_config *config,
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+ struct snd_codec *codec)
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+{
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+ unsigned int channels, rate, format;
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+
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+ if (parse_aac_header(file, &channels, &rate, &format) == 0) {
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+ fclose(file);
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+ exit(EXIT_FAILURE);
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+ };
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+ fseek(file, 0, SEEK_SET);
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+
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+ codec->id = SND_AUDIOCODEC_AAC;
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+ codec->ch_in = channels;
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+ codec->ch_out = channels;
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+ codec->sample_rate = rate;
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+ codec->bit_rate = 0;
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+ codec->rate_control = 0;
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+ codec->profile = SND_AUDIOPROFILE_AAC;
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+ codec->level = 0;
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+ codec->ch_mode = 0;
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+ codec->format = format;
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+
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+}
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void play_samples(char *name, unsigned int card, unsigned int device,
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unsigned long buffer_size, unsigned int frag,
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unsigned long codec_id, int pause_count, int pause_block,
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@@ -544,6 +751,9 @@ void play_samples(char *name, unsigned int card, unsigned int device,
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case SND_AUDIOCODEC_PCM:
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get_codec_pcm(file, &config, &codec);
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break;
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+ case SND_AUDIOCODEC_AAC:
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+ get_codec_aac(file, &config, &codec);
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+ break;
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default:
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fprintf(stderr, "codec ID %ld is not supported\n", codec_id);
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exit(EXIT_FAILURE);
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--
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2.17.1
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@ -12,6 +12,7 @@ SRC_URI = "git://git.alsa-project.org/tinycompress.git;protocol=git;branch=maste
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file://0002-cplay-Support-wave-file.patch \
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file://0003-cplay-Add-pause-feature.patch \
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file://0004-tinycompress-pass-NULL-buffer-with-0-size-to-driver.patch \
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file://0005-cplay-Support-aac-streams.patch \
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"
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SRCREV = "995f2ed91045dad8c20485ab1a64727d22cd92e5"
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